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<package pkgid="2ee2cc85f417e73338007ef038dbf1ebbe69568ec897504fc8b9228eee53f319" name="GeoIP" arch="src">
  <version epoch="0" ver="1.5.0" rel="14.el8"/>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.5.0-5" date="1370692800">- Make GeoIP.dat -&gt; GeoIP-initial.dat symlink in %install, not %post,
  and don't %ghost it
- Run geoipupdate silently in %post and cron job
- Create empty database files for %ghost to work with old rpm versions
- Don't try to use noarch subpackages on old rpm versions
- Update %description to mention database updates
- Drop outdated README.Fedora</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.5.0-6" date="1370865600">- Update sub-package requires main package for geoipupdate script</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-7" date="1384344000">- Don't download the database in %post (#1029807)
- Don't download GeoLiteASNum as it's no longer provided.</changelog>
  <changelog author="Daniel Mach &lt;dmach@redhat.com&gt; - 1.5.0-8" date="1388145600">- Mass rebuild 2013-12-27</changelog>
  <changelog author="Daniel Mach &lt;dmach@redhat.com&gt; - 1.5.0-9" date="1390564800">- Mass rebuild 2014-01-24</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-10" date="1467201600">- Added base IPv6 database to GeoIP package
- Added GeoIP-data subpackage with all other databases (#1201857)</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-11" date="1467201601">- Fixed rpmlint errors</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-12" date="1529409600">- Removed clean section
- Resolves: #1589860 - Update GeoIP legacy databases
- Resolves: #1409060 -  Missing symlink at the GeoIP-update package</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-13" date="1533038400">- Related: #1409060 -  Missing symlink at the GeoIP-update package</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-14" date="1550491200">- Resolves: #1666767 - Remove geoipupdate binary from GeoIP</changelog>
</package>
<package pkgid="8544b802b3b0d6a820e63723b54f1e273524e920dd3c65d52ee94938af89c8be" name="SDL_image" arch="src">
  <version epoch="0" ver="1.2.12" rel="23.el8"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-14" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-15" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-16" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-17" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-18" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.2.12-19" date="1518177600">- Escape macros in %changelog</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-20" date="1531396800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-21" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-22" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 1.2.12-23" date="1571313600">- Remove rpath from showimage binary to remove rpmbuild warning</changelog>
</package>
<package pkgid="814d8de91cd786b8c4a45b407c7dc44bf25a4402828689c721a40a141449bbb9" name="amiws" arch="src">
  <version epoch="0" ver="2.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
</package>
<package pkgid="b09f01e77d6cf57b40c1b852e8d5a98fffab294b43c199a954999b47d44e85fd" name="amiws" arch="src">
  <version epoch="0" ver="2.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1673697600">- Updated to release 2.3.1</changelog>
</package>
<package pkgid="be77c852088af277a6dd1c9be3e3485af878d13cec15e86606b4e5ff734865a5" name="amiws" arch="src">
  <version epoch="0" ver="2.3.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1673697600">- Updated to release 2.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.2-1" date="1678363200">- Updated to release 2.3.2</changelog>
</package>
<package pkgid="49b4c41e43cbb6883e126af02f833bf74a50ca133210d2d6c47916eceb7a355b" name="asterisk" arch="src">
  <version epoch="0" ver="16.10.0" rel="1.el8"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="eb22b743ceb5eeb8085365f5271519ec04f2d8bea80c08bfbae84ce4aac4fc9d" name="asterisk" arch="src">
  <version epoch="0" ver="16.11.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="962fcf936ceeea6049fd2af55b4b0084d06aebd72b0f253a5193421057d5ee84" name="asterisk" arch="src">
  <version epoch="0" ver="16.11.1" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="762050fca07f5dfef4ab6a69700394cee11e7c727d78ab21ce40ec3fb04da7ea" name="asterisk" arch="src">
  <version epoch="0" ver="16.12.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="85bd26114428a83dca2fc6731b02603114c1730a8cd2c38e85d30b0561e168d7" name="asterisk" arch="src">
  <version epoch="0" ver="16.13.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0a873d8bc68761e21c3535acaaba2c25fbe13408dfc38c8989dc7f6d4efb3e95" name="asterisk" arch="src">
  <version epoch="0" ver="16.14.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8a41b4f542241e50eb2e6756cb1bc73eb705f3fc09af25172a86a3395068fe2f" name="asterisk" arch="src">
  <version epoch="0" ver="16.14.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8980a3d53527bd095ffa5351f657c0a0d46a28fa998b4b05576c21088cd519ce" name="asterisk" arch="src">
  <version epoch="0" ver="16.15.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ac46e5b4c3cd166317f17a5b5cb750e07889151900f90cd25e0037b91572098f" name="asterisk" arch="src">
  <version epoch="0" ver="16.15.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="130ad8c428c8028015c959ff86a82594823dd590ce2f7e000d0c9f2448a3109a" name="asterisk" arch="src">
  <version epoch="0" ver="16.16.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b067cc7486f5ca813d6feb5e84dc08423e25aef5509beed7a07c8868920e902d" name="asterisk" arch="src">
  <version epoch="0" ver="16.16.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8e0056357c9bcdfb4fcc0afe39748ee294d8799c17189bca73dfe6b9f77cd1f6" name="asterisk" arch="src">
  <version epoch="0" ver="16.16.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1b331546733706359c5c890b73244d2789f9541f66b30896f8d2e1ca19f74312" name="asterisk" arch="src">
  <version epoch="0" ver="16.17.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d0e8ec3ece8efcdd3d5752549c2abc3446bbf116f7818cc34d5f16741ae676ed" name="asterisk" arch="src">
  <version epoch="0" ver="16.18.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="84abcde4c2898a43c11814b93fe9ea6b1c45ab8e267a4911ed31c1315d582bce" name="asterisk" arch="src">
  <version epoch="0" ver="16.19.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b70478076dcbffe61f0adbf6587161ab44356b8f09728b9c6087bda8d133fc28" name="asterisk" arch="src">
  <version epoch="0" ver="16.19.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6d09fe6d20ca511a4d06d73cb6442a0d968e18ba495f08503e0f6184d064b6eb" name="asterisk" arch="src">
  <version epoch="0" ver="16.20.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="0f5a0ac09b2b907be191a47e033476d6c2cf834e36ab8e5fd6c352fca8978e15" name="asterisk" arch="src">
  <version epoch="0" ver="16.20.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
</package>
<package pkgid="bd7ed0fa6f0f3b8ae4b19f3035a07b1cf6f4372674acc1e6247dce308c398329" name="asterisk" arch="src">
  <version epoch="0" ver="16.21.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1acc9d487d9ee0408440ad67d36a27ee442380a5dce40bda96289beddbde18b5" name="asterisk" arch="src">
  <version epoch="0" ver="16.21.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="eadfc2897ac11c73d80e4868f2cd49d1507351a1aab7fb330d985f1783210571" name="asterisk" arch="src">
  <version epoch="0" ver="16.22.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="05950b1faf108bb9e9c0f49eb9d0aaef7955246acc50c9772ccdabbe66f6a671" name="asterisk" arch="src">
  <version epoch="0" ver="16.23.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
</package>
<package pkgid="9eb321262a576435ae707f8210982e54dd6952721a20563c4922fe7bc0c2ca6d" name="asterisk" arch="src">
  <version epoch="0" ver="16.24.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ba8934cb9f6a3253c4a978be0a71d2dfe345f5df6cabaef0c7fbac1738c71106" name="asterisk" arch="src">
  <version epoch="0" ver="16.24.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9a2ce22087336c2a90bcfced9f06b642f317750aa30cff3ce8f5efb62a145f18" name="asterisk" arch="src">
  <version epoch="0" ver="16.25.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0cb5e3f2f3dbe1e21c7df6c0557d1b4b546a7b1b6e37aef34d891d2e009fcd80" name="asterisk" arch="src">
  <version epoch="0" ver="16.25.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="354cb0d7c466420f7d6fe571eade6b437a79bd431aef656a0fed0d5962b626a5" name="asterisk" arch="src">
  <version epoch="0" ver="16.25.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d0503e8d4c94fe36994c7931cb188f762f642616598b700a9048d284ec9c9679" name="asterisk" arch="src">
  <version epoch="0" ver="16.25.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6ee71931caed2e606e03d696669f089dc1a1c228b709a0e89d90396674617424" name="asterisk" arch="src">
  <version epoch="0" ver="16.26.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f89d91ae0ed329232be988e40bcbbbaa241354d508a781ed88aeb3be67e617a6" name="asterisk" arch="src">
  <version epoch="0" ver="16.26.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6f7b7fb2930030c46c537b5ef9d4ed021c83b0301af2012d910caccebbfa3870" name="asterisk" arch="src">
  <version epoch="0" ver="16.27.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="27754573d4fd988aa0b50d78efd3ac1af54bf0b04d47624ec1fca3a445baaece" name="asterisk" arch="src">
  <version epoch="0" ver="16.28.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e95ed8e0e3fcf8090b7a31485193ab19f761f42326fff5a13368709f1bd93c19" name="asterisk" arch="src">
  <version epoch="0" ver="16.28.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="68ea356aaa95380d2d3e4456e80950b763fb658b076cd1e0e871f7649609bb86" name="asterisk" arch="src">
  <version epoch="0" ver="16.28.0" rel="3.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="1394a47a6907be3f3b50018ad67bcc283b62c4a5a27b6a92ad908a4834d2e3f4" name="asterisk" arch="src">
  <version epoch="0" ver="16.29.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d77452a7035cc7cab17d85b1d874aa8152b4028f249b118d12bf877e42b4e776" name="asterisk" arch="src">
  <version epoch="0" ver="16.29.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8db93a42189370e7bd0ad2c98708d509505db1277d7d024e0fc33714b495e5f1" name="asterisk" arch="src">
  <version epoch="0" ver="16.30.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.30.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.30.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option &lt;E2&gt;&lt;80&gt;&lt;9C&gt;peer_supported&lt;E2&gt;&lt;80&gt;&lt;9D&gt;
-                    (Reported by Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn&lt;E2&gt;&lt;80&gt;&lt;99&gt;t be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: &lt;E2&gt;&lt;80&gt;&lt;A6&gt;may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5e374a6e402dcd091976a3b5759fce64d65ced27a1d5e7c55a1adc309c49c054" name="asterisk" arch="src">
  <version epoch="0" ver="16.30.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.30.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.30.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option &lt;E2&gt;&lt;80&gt;&lt;9C&gt;peer_supported&lt;E2&gt;&lt;80&gt;&lt;9D&gt;
-                    (Reported by Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn&lt;E2&gt;&lt;80&gt;&lt;99&gt;t be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: &lt;E2&gt;&lt;80&gt;&lt;A6&gt;may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 16.30.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/16.30.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 16.30.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.30.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/16.30.0...16.30.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-16.30.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list
-              before applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="e2b51806cbbef0c6c1ca6db585d1d00b5f67d7c77c059736f599356a6db7f2fe" name="asterisk" arch="src">
  <version epoch="0" ver="16.6.1" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.0-1" date="1550232000">- Update to upstream 16.2.0 release for bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.1-1" date="1551441600">- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
</package>
<package pkgid="bb2091165a33f81bac467a04a5f3daac4b11f709402c8d9487902e1a978d80aa" name="asterisk" arch="src">
  <version epoch="0" ver="16.7.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bd094e15807ede9554982e53e84ed5aa0b96ab6d9f10d0e1548c1cc011759206" name="asterisk" arch="src">
  <version epoch="0" ver="16.8.0" rel="1.el8"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1bf31403350343d5d2e51e375af34da6d5122e5180566ddabf54c10e3d717f7f" name="asterisk" arch="src">
  <version epoch="0" ver="16.9.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl) 
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes) 
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b5c62b7a5996e315c319907630b25cac432bfc3646808ced3a2afc7772c82b7c" name="asterisk" arch="src">
  <version epoch="0" ver="17.0.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.1-1" date="1551441600">- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b29598e7e45afda50b9985d9bad2efd876bc70adea7aee98649c03c0e3c2d691" name="asterisk" arch="src">
  <version epoch="0" ver="17.0.1" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="dd1fdf2d150f86ca0ae3977c566898496a270661724c58fd60d135b4c7fb6b05" name="asterisk" arch="src">
  <version epoch="0" ver="17.1.0" rel="1.el8"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d057227623663aaad4ebd86fb71484897d182856a53f20940578d425f92cf5ad" name="asterisk" arch="src">
  <version epoch="0" ver="17.2.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8ba437899bcf530eb247100aa3164ab8191aa77686e3c38f7ec6e3172e76fd22" name="asterisk" arch="src">
  <version epoch="0" ver="17.3.0" rel="1.el8"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="02b430f37174b00a55323106086426e69a445a25f798f23a160006b2dca6b1ad" name="asterisk" arch="src">
  <version epoch="0" ver="17.4.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="88c23fba78b8c97854665c39523110c8e7a7adcf05013945dcd5538000139a9f" name="asterisk" arch="src">
  <version epoch="0" ver="17.5.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1e40b09543d4933102f9048465c7b84d9a1326890a41030f6d4bfd8e71f9a8fa" name="asterisk" arch="src">
  <version epoch="0" ver="17.5.1" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="97bfcab5f5bfb8bebd8d892601ba67bad0a382402b724fdcd5107ddb1e1e0f39" name="asterisk" arch="src">
  <version epoch="0" ver="17.6.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0409edef66731d023b4628d00fadee407a0d20c9a3be318a349cd4cf982eb69f" name="asterisk" arch="src">
  <version epoch="0" ver="17.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ab519bc24d288b2ae2fb68afdf40ea08f11f3b084abc36f0f9c8e81d0106b6df" name="asterisk" arch="src">
  <version epoch="0" ver="17.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="92a74c30e80854d31c16eaf8dee1249748765de671b2ff7d7ff6421a1ec93e42" name="asterisk" arch="src">
  <version epoch="0" ver="17.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1bacd05f3ae5d3094b864d448b7874ecb3f7e239f5ce17fa75068d288cdec649" name="asterisk" arch="src">
  <version epoch="0" ver="17.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7783e2e59e15c28377b74a8ed6a09b47cf383e65d3f4828ba78b4bf72958b816" name="asterisk" arch="src">
  <version epoch="0" ver="17.9.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9c5130549944a8d5e225bc6586602c61be2f856f2b75fd781cbf89279841fb2c" name="asterisk" arch="src">
  <version epoch="0" ver="17.9.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="46915834fb95f641be618a120f60a2ed1944628d912579b517d4aa8cd3566c87" name="asterisk" arch="src">
  <version epoch="0" ver="17.9.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="12d182c687f93df3f36ffa51dd737fa378ed57eb769cbe487aad5e3924168905" name="asterisk" arch="src">
  <version epoch="0" ver="17.9.4" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3a3b12366558889f341de513ece7135f2f6f2feb376c286bad8d6cb257cdc15d" name="asterisk" arch="src">
  <version epoch="0" ver="18.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c4592397fbc5f2e1b126fedc911e88f80be95e5e1472255c08dd8d493d56dd39" name="asterisk" arch="src">
  <version epoch="0" ver="18.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e5a724535035ea1264c4fcb6efa4d173552551f11708678093c5ede104168442" name="asterisk" arch="src">
  <version epoch="0" ver="18.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4fa068bd579ab2b90e540600fcafdc6362737d9d0fc8ab6277bbe4065ecd2037" name="asterisk" arch="src">
  <version epoch="0" ver="18.1.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7909339c342b9b320f7ae875b4cdb4fa0523c4860acdc0eb77c5430da6641e7a" name="asterisk" arch="src">
  <version epoch="0" ver="18.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7222616a330d5d8acfcad51865813a2ed2f725c1a5ad9bb4bc37138b543cf78f" name="asterisk" arch="src">
  <version epoch="0" ver="18.10.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="49b505e6644464f4d92a5e2c0c8393c5eaa0612efe82e4fd8c488dfe58292d42" name="asterisk" arch="src">
  <version epoch="0" ver="18.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a3a61670841b10e2db3aba14809489cc05c41e2ad05025c265c24d567cd50e20" name="asterisk" arch="src">
  <version epoch="0" ver="18.11.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1aef592d2730ea046e1b59777eb3cf11cfb12048c63da5b40f0ff157fb736de3" name="asterisk" arch="src">
  <version epoch="0" ver="18.11.1" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="cd4cff011e2bd20c39c8f588fe6179fd9ad5a1778d69659650484b8c52b25557" name="asterisk" arch="src">
  <version epoch="0" ver="18.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b8f77afadec74d5ab5f07895233351ce9f8703fbbc4800675109c896b8e5ab94" name="asterisk" arch="src">
  <version epoch="0" ver="18.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9b38e87de8f33bdc2779c29d8f3a56390438b624272d5df768a88ab940acc82f" name="asterisk" arch="src">
  <version epoch="0" ver="18.2.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0a79a66cdb2dfa46cc9c036a9a5bf090dc6a6299f6b9a30992806c549983ac1f" name="asterisk" arch="src">
  <version epoch="0" ver="18.23.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="443183fb53b7ce45387f9a26663d81bb9a7c6b9e19efd45a983570c85d7cd915" name="asterisk" arch="src">
  <version epoch="0" ver="18.23.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="a9242b0fd21e2b92c107d295d4666c83c8ce749176463d43846a048114636b92" name="asterisk" arch="src">
  <version epoch="0" ver="18.24.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="0fb5e73ad1b221beabfe1181a082cddb2ea6e3dae645aa290ec81502215690e0" name="asterisk" arch="src">
  <version epoch="0" ver="18.24.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="73d95e4248814b7163551084fd4da8fb471c2ddd9e9d50826fc4e62411cb5a97" name="asterisk" arch="src">
  <version epoch="0" ver="18.24.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="d2ff63820eb30e38114aefdc54a6cb52c31d6992d8d722777f903e913d2a0abe" name="asterisk" arch="src">
  <version epoch="0" ver="18.24.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="407e2cec159b24b515c885b5c6872a8e310b39a6faf88e0fd0bd4a850878e569" name="asterisk" arch="src">
  <version epoch="0" ver="18.25.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="2dbb33e08bb75a69226a9f04a0c3ccf496cdca5d12655644ee59b9ed72438804" name="asterisk" arch="src">
  <version epoch="0" ver="18.26.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
</package>
<package pkgid="e43890306cbbdac641038cfa6ee711594a803a8983705e884f2ee0510921c3e9" name="asterisk" arch="src">
  <version epoch="0" ver="18.26.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="9eb340e553d6be5b7105fcc04529aad59e3d8a64f5e46f157a9db16844b4c9e1" name="asterisk" arch="src">
  <version epoch="0" ver="18.26.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="ed1b2fde7ac1a61e4affd7787fa3fd99f5e67a8062a41014c10cad11b25f1b0e" name="asterisk" arch="src">
  <version epoch="0" ver="18.26.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

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

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="550376e5347f185dda665defb6b77fbea50f2bd4a01535001b3100148161c754" name="asterisk" arch="src">
  <version epoch="0" ver="18.26.4" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

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

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.4-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.4.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.4 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.4
-
- Change Log for Release asterisk-18.26.4
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.4.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.3...18.26.4)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.4.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-557q-795j-wfx2](https://github.com/asterisk/asterisk/security/advisories/GHSA-557q-795j-wfx2):
-    Resource exhaustion (DoS) vulnerability: remotely exploitable leak of RTP
-    UDP ports and internal resources
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-557q-795j-wfx2: Resource exhaustion (DoS) vulnerability: remotely
-                       exploitable leak of RTP UDP ports and internal resources
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    pjproject: Update bundled to 2.15.1.
-
- Commit List:
- ------------
-
- pjproject: Update bundled to 2.15.1.
-
- Commit Details:
- ---------------
-
- pjproject: Update bundled to 2.15.1.
-            Author: George Joseph
-            Date:   2025-08-25
-
-            This resolves a security issue where RTP ports weren't being
-            released causing possible resource exhaustion issues.
-
-            Resolves: #GHSA-557q-795j-wfx2</changelog>
</package>
<package pkgid="7fc0d17ccc3d9af9dddd95cbe8e1f45502d5b4079ee844cec6538d0033d8b7a6" name="asterisk" arch="src">
  <version epoch="0" ver="18.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bfa13542577d7e5a81d1d87f2d0cf075bcae727e569209703d03580a0ebc0cca" name="asterisk" arch="src">
  <version epoch="0" ver="18.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="264e062922bca06f6710bbba3ed510ab5d04081cd86c528b673031d3e61463e4" name="asterisk" arch="src">
  <version epoch="0" ver="18.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="49f9db1d8c7a642ca73e6ae042c20c7900607f7593629b367372361431349a2e" name="asterisk" arch="src">
  <version epoch="0" ver="18.5.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="189ce60189f2bfdc00a1a33df540f1066f38b00d2982e688927233cf3048de8c" name="asterisk" arch="src">
  <version epoch="0" ver="18.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="bd330201a2fe80b371468dea1c98976183944f89367d59d6d84709c0a843b8b9" name="asterisk" arch="src">
  <version epoch="0" ver="18.6.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
</package>
<package pkgid="9990a59eed6604507b653907f8b8f9fe91cb56438c0eca447feb4314cc35b13d" name="asterisk" arch="src">
  <version epoch="0" ver="18.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7e2038861a52db8a421922a6f032feac0d8757fbe7635e1682cd06fba2bbc1ec" name="asterisk" arch="src">
  <version epoch="0" ver="18.7.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0b673e43ecce03c702a7be1713eef479aab1e33724b216755068502fba02ebc2" name="asterisk" arch="src">
  <version epoch="0" ver="18.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e3a08c2058dc918f0a9e4731d055aef3a826cbbaa25575f8f9b8ed4e28f52db8" name="asterisk" arch="src">
  <version epoch="0" ver="18.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6ce79e6391170ffb755c8d73b9e90bb42e5aa207218fab16030dab2b1f4576e2" name="asterisk" arch="src">
  <version epoch="0" ver="19.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3c749f8a68d4e324dd84812cefb520d8390b5f3580bbaa8dc442d8f3a9d256a9" name="asterisk" arch="src">
  <version epoch="0" ver="19.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7d8a150c775369cf3c08516c9e0a5574c4a64f8bb323001c358e9e04b431737c" name="asterisk" arch="src">
  <version epoch="0" ver="19.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d0bd434d5c1991d850b5066c67b17817728b946f97ecc4dcbdb58fb8666ff8c2" name="asterisk" arch="src">
  <version epoch="0" ver="19.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="81fb5bc4f3f8edb284ea780c4580c2f1c1ede8e0f7efc232bcc9023e1a576abb" name="asterisk" arch="src">
  <version epoch="0" ver="19.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4d47d237e2d9d7e8745ac3a0b3887223eca81758c0651474bca18f05212d8595" name="asterisk" arch="src">
  <version epoch="0" ver="19.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3d533e246c852cf7c052589bff64c5fe6b976e5f816d325518408b4c5c0c1d28" name="asterisk" arch="src">
  <version epoch="0" ver="19.3.1" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="587db4ad15b871722fb3acbcc7bf4e68cfcd2c7243c03593fe190d1a06d47a07" name="asterisk" arch="src">
  <version epoch="0" ver="19.3.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ab6a1870ac4cdc13bb995df52f05c0f143379c1983dfd1ab8d641099558cd021" name="asterisk" arch="src">
  <version epoch="0" ver="19.3.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c4300ea22a7ca62525ca0fcf4e98bb5020266c37c672b644ef2fcf7ad5c6fb95" name="asterisk" arch="src">
  <version epoch="0" ver="19.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f6c6b099b905c03d302acb033b00372029d2d1ab1ccc814053f97e6ba19ef1d4" name="asterisk" arch="src">
  <version epoch="0" ver="19.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c0279ba3a33338472b5c369bcf1a8aff07127b7e40a6ecd027b80fd9a2674199" name="asterisk" arch="src">
  <version epoch="0" ver="19.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a8d02fde52bd5a556bb04a7eb0188f0d044eb7e17103cd35090bba08a2e10e4a" name="asterisk" arch="src">
  <version epoch="0" ver="19.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6d732b700c3c9e2ed8d2efc2a91e7083679ab735f8162a38797bc6f451f6bbc7" name="asterisk" arch="src">
  <version epoch="0" ver="19.6.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="fd5f98992201da1767b48f5843d71a68b655c08401f1d49baa831d9a043904bc" name="asterisk" arch="src">
  <version epoch="0" ver="19.6.0" rel="3.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="d424134e684bbfefe5db1b510f07f6469490685143fe95f8e3df4c785e5c58b3" name="asterisk" arch="src">
  <version epoch="0" ver="19.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="68dc4ba98c5697f5a6d85a4cc47df6e565a3da37137dcddd3e89e114dedc2245" name="asterisk" arch="src">
  <version epoch="0" ver="19.7.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="452c9acbe72eee7da96c12980086f5e9e89b14a436b52bd04e1bafcdb672f22c" name="asterisk" arch="src">
  <version epoch="0" ver="19.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896001">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="96ca9178e05527b56ad1de6f8ed3fcf8402c822d88294e3bb631e4f347e12ed8" name="asterisk" arch="src">
  <version epoch="0" ver="19.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896001">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 19.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/19.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 19.8.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/19.8.0...19.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-19.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="36bc1c4f7ffde2c77cb363c5ee7f553b08b03e95100c05e0bacec3b156b6e70f" name="asterisk" arch="src">
  <version epoch="0" ver="20.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4038978bf650e0de0b8ab78de1740fd9c8754cb4f8140fb29cde5aa1d2cd535c" name="asterisk" arch="src">
  <version epoch="0" ver="20.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c2c9462ee155bb8b05e202edcb57e8d7dbb2ab2ea7cec1ef2c33e1b0ae6c6a0d" name="asterisk" arch="src">
  <version epoch="0" ver="20.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a75894457dce61fa5731367c19cf04ae1a3ce6209381bc15c3fbcc2ad42636da" name="asterisk" arch="src">
  <version epoch="0" ver="20.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="7d521731f80b5328958e2c7afa4e19a705b39a06b95fb7239275b687e1d3003a" name="asterisk" arch="src">
  <version epoch="0" ver="20.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="117b82a03e6de4530da07a92ffba55bbb670a47933658d56da028a1fd13e24b9" name="asterisk" arch="src">
  <version epoch="0" ver="20.11.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="71c626ba8200b88d28f3435867d14d60b0c03590dde2e4a7c3e4a76b119c5d2d" name="asterisk" arch="src">
  <version epoch="0" ver="20.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="f43b0589e7e1986ebab8ed81b211621065e98a65fa1f1703f8c1594302418c52" name="asterisk" arch="src">
  <version epoch="0" ver="20.13.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="81699e0b274906e85973e03d7d9138344c0f6d4175533ff1ce16ac66ee39e654" name="asterisk" arch="src">
  <version epoch="0" ver="20.14.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="7436533a7d40108d3a4db088f3e905a97a33ae71ef24113d748dbc9abf23e2de" name="asterisk" arch="src">
  <version epoch="0" ver="20.14.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="76387e1514e6686658ec132fae76e607b716be9f54dab33ec475f28200d43fe4" name="asterisk" arch="src">
  <version epoch="0" ver="20.15.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="ba919f0f1ed72e2da2b17954cd0ac2b1ea0750d1ab01a401c9859e4b0ca87854" name="asterisk" arch="src">
  <version epoch="0" ver="20.15.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="cc1f5c07e91916d523cab3cbc3728f13a95b59c1c9037e7fa933528cfeda292c" name="asterisk" arch="src">
  <version epoch="0" ver="20.15.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="fbc6c177c67b6c380b1218f5c1957f3f142741d988a0d056395964bbb6e9f006" name="asterisk" arch="src">
  <version epoch="0" ver="20.16.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="058c41f5fcffacc33898e00a9724d76bfd4df8ac1c723baf87f06713a67c3b30" name="asterisk" arch="src">
  <version epoch="0" ver="20.17.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="9591217c7d6982c023d2be23c32d4ce353d3361f58901362d9111f613183fdf7" name="asterisk" arch="src">
  <version epoch="0" ver="20.18.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="71dc65541576200dbfc5c1ef366f059d76083308641c448ac169937ae79013a1" name="asterisk" arch="src">
  <version epoch="0" ver="20.18.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="21e14169142ff61e49622e55cfd41040a721a049da6c7249d71f0eb23c32262b" name="asterisk" arch="src">
  <version epoch="0" ver="20.18.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="895b379b7ebe2a5511c50d1ef3ab660082c1c801204d29b381cf96d62526fa50" name="asterisk" arch="src">
  <version epoch="0" ver="20.19.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="d76f261d457336370fe7aa92777f30f94aab4437a14883d4f4a23c42f2b2e5ef" name="asterisk" arch="src">
  <version epoch="0" ver="20.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ed8edd90801b2e15a8461f76bd58f0089c4c6ca012e7151a78d2e3b0f00be7e4" name="asterisk" arch="src">
  <version epoch="0" ver="20.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fa8a9b85d2f713cd0bb2924f40ad6ca28059e4ccb032c447b13613fdd1cde079" name="asterisk" arch="src">
  <version epoch="0" ver="20.20.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="5bf2bd3f6ea3c673c44703a5e3e1471ee65d0755c9341e3690d4173aa296dc0f" name="asterisk" arch="src">
  <version epoch="0" ver="20.20.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="cb7fdfbe85c678785df744ab28663e2b0cef2b5ff6e58b6eeadb0a55931053dc" name="asterisk" arch="src">
  <version epoch="0" ver="20.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
</package>
<package pkgid="a6a423bfec5e49b6d4c614f329c0a606b0ee2c106fc7c7cd183e40a98cdc67f6" name="asterisk" arch="src">
  <version epoch="0" ver="20.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="3b89a389661d087301879a99000b3e4eb349013dff39769cab156c11e0723342" name="asterisk" arch="src">
  <version epoch="0" ver="20.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="2d5d7b2f7b04c1d0babb11414f48fd43c654358920c4a6b846d63c24037d0a62" name="asterisk" arch="src">
  <version epoch="0" ver="20.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="7beede150d4ce4b873f038704e475dc8536be1add5cabb430ae4922a7dd12ee5" name="asterisk" arch="src">
  <version epoch="0" ver="20.5.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="8c25facbcd0c491e8107eedbdd5178dfeaf06ffe3403ed03f357219775443f72" name="asterisk" arch="src">
  <version epoch="0" ver="20.5.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="b117f8c93fa3e8b729195b708b66a28b563b3fc7db14a80f572315ac83b398a4" name="asterisk" arch="src">
  <version epoch="0" ver="20.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="5714ab5cc70b0010e48d5f25bb1e9fd5605c8ec3833b0a95d83024ca142bfd30" name="asterisk" arch="src">
  <version epoch="0" ver="20.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="680b7872e101809b6456a5dd440be156d87922a742888bf81ccdd174cb301e45" name="asterisk" arch="src">
  <version epoch="0" ver="20.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="4ecbedf5436d5972accbc3f05056e0387421f4fce995260399914d3c0a2a997f" name="asterisk" arch="src">
  <version epoch="0" ver="20.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="a421ff6851cacce2fc5a59af4c046b8c104bb1ddb804f728552a8bd26ede527d" name="asterisk" arch="src">
  <version epoch="0" ver="20.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="b23dd0e391ad16e0d61c68cd6df6b8ff663cc9706a4a9314cd12fe803637500a" name="asterisk" arch="src">
  <version epoch="0" ver="20.9.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="aa6b25137019d9c223a47bd5b89c68af423025a38d2ad0b33f3ecee5d78c7646" name="asterisk" arch="src">
  <version epoch="0" ver="20.9.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="b71e15fb84349bd53856f9a48647c293182a155174f31225458d59b5c0ef3159" name="asterisk" arch="src">
  <version epoch="0" ver="20.9.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="69dedd607afb278eaf89a2163ab299a782667fde1b51259f5abde81c8ddfced0" name="asterisk" arch="src">
  <version epoch="0" ver="21.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="aef497748bb23efcea897a14eb645c2f4023f89a12aaec0d1765eb19432a79e5" name="asterisk" arch="src">
  <version epoch="0" ver="21.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="dc8d6ed45eeed2ea0fedf99054f6e080d4f2b957050aee1d9ac86a475e806a8a" name="asterisk" arch="src">
  <version epoch="0" ver="21.0.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="6be6590b1e4dd356350c2c4b942c099671039c8f02954652ad9ade5ecdb339ab" name="asterisk" arch="src">
  <version epoch="0" ver="21.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="f593ebb0d626df10a800442c8ab569904bce5c85e36d2b7ea567a056f375b39a" name="asterisk" arch="src">
  <version epoch="0" ver="21.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1747915201">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="907cca046186d4ba76e3cb362e040a9d8159db94d1af188a4a5096a238c170c3" name="asterisk" arch="src">
  <version epoch="0" ver="21.10.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="9d69eb65293be624e0a44e80d6bf5915bc5b859ba4c97d1cbeda1ed3967b32e7" name="asterisk" arch="src">
  <version epoch="0" ver="21.10.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="b3081cea6174f9f9e4ac1a380946e9524ec54bf8d1d6480ab57bb8ad48af0833" name="asterisk" arch="src">
  <version epoch="0" ver="21.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="1d7387d44de3e276d4c169b961f2f334ea500a2dbcb1a0177950f89ab435358a" name="asterisk" arch="src">
  <version epoch="0" ver="21.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="322db3ef656c476f0e3280e4614edf92ae07b891c35fcf265f92290df12342de" name="asterisk" arch="src">
  <version epoch="0" ver="21.12.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="a1dce3333f63db81aabb51576b56478a2e8d27fb8ef6e68073e5a1b228010e50" name="asterisk" arch="src">
  <version epoch="0" ver="21.12.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="4d9849401b0eeef7087e6fa7c8b962641a5a11da76b7f48932cb332c74b8ad7b" name="asterisk" arch="src">
  <version epoch="0" ver="21.12.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="dad1592c14ba40ac14babe6aefbd1e5cdc7f142687628a094ec2cb4e57835862" name="asterisk" arch="src">
  <version epoch="0" ver="21.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="0462b9327936d491f490e52c2c80eb35eb2fedad4f3d84d83a45c13669c78c1b" name="asterisk" arch="src">
  <version epoch="0" ver="21.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="9ae506094c8773fa16d2ab8ece7727b017912934d7cf04c010d3c60ca4112109" name="asterisk" arch="src">
  <version epoch="0" ver="21.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="b0153b0f3948922a6635612e624efe917f929c6ca4ad75d26f07bece200d0960" name="asterisk" arch="src">
  <version epoch="0" ver="21.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="288bb317e5b63599a3d903496e936ad0c3cdf3e8379cfc1c126c32d97c3e9309" name="asterisk" arch="src">
  <version epoch="0" ver="21.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="4aca7c69005514c51500d90b10acf2bc04a86855409e659bda07d4fdbcf19d63" name="asterisk" arch="src">
  <version epoch="0" ver="21.4.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="2bfed07d2a2d588b81fbf86d982b39f42957aac58cffe9667b737ee2dc20fb43" name="asterisk" arch="src">
  <version epoch="0" ver="21.4.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="a98680fdb292420e4650d327ddd6f847bb170c71b76e42fcd90279ff3e9507ab" name="asterisk" arch="src">
  <version epoch="0" ver="21.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="9a6fac72d2e59a3b9e9f1864e4e78631d3a137daef2b673475f86ed32cd84531" name="asterisk" arch="src">
  <version epoch="0" ver="21.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="528b0350c26b8e9f87a81885f3058d00ccbe02e65c4b4997435b479a3ac7d264" name="asterisk" arch="src">
  <version epoch="0" ver="21.6.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="27bf834a385c0502e1170d7f8b4543a8f634f27bcefcdb3b9d7a32cb8d175a43" name="asterisk" arch="src">
  <version epoch="0" ver="21.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="45af750d91aab1c7f29083e1ac1902528a53822f3820a87bf8d80e49042eb8f9" name="asterisk" arch="src">
  <version epoch="0" ver="21.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="2f0de2dcc7b3d9bedf58a673c5e01e2f3ed771aea1698e3b930f51ee00d11063" name="asterisk" arch="src">
  <version epoch="0" ver="21.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="82f395fcfc8c64b04295910c1b63546fe1ff0f5583d7ac013fd18fe1c6fb2d62" name="asterisk" arch="src">
  <version epoch="0" ver="21.9.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="b2e799d86a996343de7215ee6ee4ac14fe2758f1f0a920e99be89c27c6301275" name="asterisk" arch="src">
  <version epoch="0" ver="22.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="1c7f6beddefe2b296d16523f22787c9c0c34a5d6cb480e144f5b53cc4fcfa39d" name="asterisk" arch="src">
  <version epoch="0" ver="22.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="6868a6b784ec4ab5eed9a98ba725a984f8cb5eea168ba03948600797b93b68f7" name="asterisk" arch="src">
  <version epoch="0" ver="22.1.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="fb06ba3a4e669620aae7b624586e311cd8736140c7405f52190913808d6f7251" name="asterisk" arch="src">
  <version epoch="0" ver="22.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="98c7c29d0af0a33b5e2c817328b602bce0d75e048f770e6034da64f7f07feebc" name="asterisk" arch="src">
  <version epoch="0" ver="22.10.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk 22.10.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.1
-
- Change Log for Release asterisk-22.10.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.10.0...22.10.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="730ac4412a5185f1c6bd1a8305973fd64826027996976614a8456f7ca2252fea" name="asterisk" arch="src">
  <version epoch="0" ver="22.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="f756f174a6f460f121e3ba2686064cd0dab4cd8c4466f0439191aeddc05a2d37" name="asterisk" arch="src">
  <version epoch="0" ver="22.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="f012769d59ead1a2ac54ec05935fa0e7e46f6769521eac234b7c94308a2d85dd" name="asterisk" arch="src">
  <version epoch="0" ver="22.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="bf875a5a4a8f76a65d88aa91080d836d472ff058100870f3caa34a382c69a0f2" name="asterisk" arch="src">
  <version epoch="0" ver="22.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="c6d4931d3e4c52f82a6e972fd3e8d97a553130b2f2e91a8f5e259bed19946d78" name="asterisk" arch="src">
  <version epoch="0" ver="22.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="88ee871f234908a6506cd7dd874daf78164d2ebcec8f77787b3fe51ca136f162" name="asterisk" arch="src">
  <version epoch="0" ver="22.5.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="0fb03037d13ec7c5be5060ee6fd67cfcd7245e1fad54dc6dda3e8efb5ad513c1" name="asterisk" arch="src">
  <version epoch="0" ver="22.5.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="9bf02f9a4eb5bc06edc60c4da67a253926e8a46cfaf1856eee18facee6462525" name="asterisk" arch="src">
  <version epoch="0" ver="22.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-            The new "suppress_moh_on_sendonly" endpoint option can be used
-            to prevent playing MOH back to a caller if the remote end sends
-            "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- app_mixmonitor: Add 'D' option for dual-channel audio.
-                 The MixMonitor application now has a new 'D' option which
-                 interleaves the recorded audio in the output frames. This
-                 allows for stereo recording output with one channel being
-                 the transmitted audio and the other being the received
-                 audio. The 't' and 't' options are compatible with this.
-
- manager.c: Restrict ModuleLoad to the configured modules directory.
-            The ModuleLoad AMI action now restricts modules to the
-            configured modules directory.
-
- manager: Enhance event filtering for performance
-          You can now perform more granular filtering on events in
-          manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-          = PJSIP/` This is much more efficient than `eventfilter = Event:
-          Newchannel.*Channel: PJSIP/` Full syntax guide is in
-          configs/samples/manager.conf.sample.
-
- db.c: Remove limit on family/key length
-       The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-       removed and will now accept families and keys with a total length of
-       up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-       dialplan applications, dialplan functions, manager actions and
-       `databse` CLI commands. Since the media_cache also uses the
-       `ast_db_*()` APIs, you can now store resources with URIs longer than
-       253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="fed154538faccb760f5e99cbd8980e8b2c36fb5ef30c93e1de34b544a2b277c3" name="asterisk" arch="src">
  <version epoch="0" ver="22.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="914f859af4b5192dd00dc953dd8859ad18e22b2d73c64f02960fd37c99b816ee" name="asterisk" arch="src">
  <version epoch="0" ver="22.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="975f50ecc4e69bf3c05eec75784faa2c2dcf99762324db805de9ba82b17015e5" name="asterisk" arch="src">
  <version epoch="0" ver="22.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="7df6f329d91360fe8b60d3b95a5399af583155c7540eb846bcb69099ef8f13b3" name="asterisk" arch="src">
  <version epoch="0" ver="22.8.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="4bf462784d807627bff4db89dbf1c6ce30035c17099ffb5c9c856de5f7702a82" name="asterisk" arch="src">
  <version epoch="0" ver="22.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="9853bf36dc878bbb21392bac1fd8ae7cd917db333ac550dc122aa4118e22e06f" name="asterisk" arch="src">
  <version epoch="0" ver="23.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
</package>
<package pkgid="1fd214bf9159c5317823d5a05d6e6500899395d4377d669fa732be3207a4871c" name="asterisk" arch="src">
  <version epoch="0" ver="23.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="445a5104003ee3b78246fa359ff56ebcf304dc8e86f282eb9ea1293f3fc6a52e" name="asterisk" arch="src">
  <version epoch="0" ver="23.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="3b6879c1ada57a003933d0541de97e25ff6038e27e3f20d4bc2fb12e40302b1f" name="asterisk" arch="src">
  <version epoch="0" ver="23.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="45a7ec09ffaeaec16b0559afe2ac43ccf4c7e30c958e8e8941561077ced02053" name="asterisk" arch="src">
  <version epoch="0" ver="23.2.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="36103f15a6bce53695085ccdaf04c6d597528ab2402b16facdfbd4a7dec77d0d" name="asterisk" arch="src">
  <version epoch="0" ver="23.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="53b79a222ad0dd572673ef13de83f74852b637e86a1a199fec453af0e173a785" name="asterisk" arch="src">
  <version epoch="0" ver="23.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="72f386681d7c287b04dfda34f79e0f60b88a127fa4e50c48a2730a238696d844" name="asterisk" arch="src">
  <version epoch="0" ver="23.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.1
-
- Change Log for Release asterisk-23.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.4.0...23.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- #### ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="5bc61adcc492540a3f23adf46814f4c84177969d2475c3fe37cbaf07bde440da" name="asterisk-chan-dongle" arch="src">
  <version epoch="0" ver="1.1.r44" rel="2.el8"/>
  <changelog author="aevseev@gmail.com - 1.1.r34-1" date="1423656000">- Build from GitHub repository of FiW
  + Add patch to support for send SMS to short (non-free?) numbers
  + Add patch with DongleSendUSSD application</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.r44-1" date="1530705600">- Upgraded to commit 44
  + Add 61 patches up to 2018</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.r44-1" date="1530705601">- Added asterisk to dialout group
- Fixed CentOS/RHEL 7 tmpfiles lock system</changelog>
</package>
<package pkgid="98ca0b972f97731b7048698bd71a218baa28a7cb00fd230d6fdd376285cf75ad" name="asterisk-codec-opus" arch="src">
  <version epoch="0" ver="1.3.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.0-1" date="1515844800">- First build script</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.0-2" date="1539172800">- Updated to asterisk 16 support</changelog>
</package>
<package pkgid="0fabb87c60ce65be46aea9b6dc68b5c354cda3af8b0ace6afd22ccfd1bb526b9" name="asterisk-codec-silk" arch="src">
  <version epoch="0" ver="1.0.3" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.0.3-1" date="1515844800">- First build script</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.0.3-2" date="1539172800">- Updated to asterisk 16 support</changelog>
</package>
<package pkgid="7eac9d0cd218b4c502ee63f65aab2eb1a677c0a5b29f4882c625feb53a6bec24" name="asterisk-codec-siren14" arch="src">
  <version epoch="0" ver="1.0.7" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.0.7-1" date="1515844800">- First build script</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.0.7-2" date="1539172800">- Updated to asterisk 16 support</changelog>
</package>
<package pkgid="1c46cef0cfccf03741e5629d102d0fa97f95af4f9688a5c88ba2034203b116f1" name="asterisk-codec-siren7" arch="src">
  <version epoch="0" ver="1.0.7" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.0.7-1" date="1515844800">- First build script</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.0.7-2" date="1539172800">- Updated to asterisk 16 support</changelog>
</package>
<package pkgid="960ace70bb809720b03a108358d4b13e7abffe7f3ece723a01011f1ac2b04440" name="asterisk-sounds-core" arch="src">
  <version epoch="0" ver="1.6.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.25-1" date="1400068800">- Spec base on asterisk rpm packages</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.26-1" date="1410004800">- Update version to 1.4.26
- Added japanase language (ja)
- Added g729 siren14 siren7 sln16 wav formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.27-1" date="1431000000">- Update version to 1.4.27
- Added slovenian language (sv)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.5-1" date="1452945600">- Update version to 1.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6-1" date="1499947200">- Update version to 1.6
- Added New Zealand English (en_NZ)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.1-1" date="1516968000">- Update version to 1.6.1</changelog>
</package>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.11.19-1" date="1570708800">- First version of this package</changelog>
</package>
<package pkgid="27fd3edddddb2b4174affdb642f4fa1e7a3873e6bb056743f1ac8995991d840e" name="asterisk-sounds-core-es_AR" arch="src">
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.11.3-1" date="1570708800">- First version of this package</changelog>
</package>
<package pkgid="be1bc4e14dab992068ad0a163ad398d5a429f0115808f3a512197b268e28590c" name="asterisk-sounds-core-es_ES" arch="src">
  <version epoch="0" ver="2.9.15" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9.15-1" date="1570708800">- First version of this package</changelog>
</package>
<package pkgid="a0e280fd5032672fceeace6741008a11807097fa095e9dac1dda5e2ff3b6c8b3" name="asterisk-sounds-core-es_MX" arch="src">
  <version epoch="0" ver="1.11.11" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.11.11-1" date="1570708800">- First version of this package</changelog>
</package>
<package pkgid="29bc8030b9c71dc747e7d61c3d3ccee0accac0873259ad377790140c1461a0b7" name="asterisk-sounds-core-fr_FR" arch="src">
  <version epoch="0" ver="2.3.10" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.10-1" date="1570708800">- First version of this package</changelog>
</package>
<package pkgid="b8b839dd8181822c031ff00e7eebd5b31f2c72fe90135b1849342a617c6a1f6f" name="asterisk-sounds-core-it" arch="src">
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</package>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.8.3-1" date="1529496000">- First version of this package</changelog>
</package>
<package pkgid="f49d37c877b11a6e70143ba58608eed8ca464b45cdbe841572d5cefc257137e6" name="asterisk-sounds-extra" arch="src">
  <version epoch="0" ver="1.5.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1571140800">- First version of this script</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.11-1" date="1571140802">- Upgrade to version 1.4.11</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.12-1" date="1571140803">- Upgrade to version 1.4.12</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.13-1" date="1571140804">- Upgrade to version 1.4.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.14-1" date="1571140805">- Upgrade to version 1.4.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.15-1" date="1571140806">- Upgrade to version 1.4.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.5-1" date="1571140807">- Upgrade to version 1.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.5.1-1" date="1571140808">- Upgrade to version 1.5.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.5.2-1" date="1571140809">- Upgrade to version 1.5.2</changelog>
</package>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.11.3-1" date="1570795200">- First version of this package</changelog>
</package>
<package pkgid="a14803dbc32bc247ec0104c10ce7aeffe08f34422b13b30750d249d41509be60" name="asterisk-sounds-extra-es_ES" arch="src">
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9.15-1" date="1570795200">- First version of this package</changelog>
</package>
<package pkgid="e58d774a821db4b67421d82c03999344993aeb8694ee26d10d8b289b64809308" name="asterisk-sounds-extra-es_MX" arch="src">
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.11-1" date="1570795200">- First version of this package</changelog>
</package>
<package pkgid="8e40fe708d9c9ab08f2e64b84ca3ab1648f0bdc4a9ca705903f897574d6631bd" name="asterisk-sounds-extra-fr_FR" arch="src">
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.10-1" date="1570795200">- First version of this package</changelog>
</package>
<package pkgid="e40469e8eb1e6cf9a34f9416d4d23e082d0d0d81fe26404384bc21dc6e7554b0" name="asterisk-sounds-extra-it" arch="src">
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9.13-1" date="1570795200">- First version of this package</changelog>
</package>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.11.10-1" date="1570795200">- First version of this package</changelog>
</package>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.01-1" date="1571140800">- First version of this script</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.02-1" date="1571140801">- Upgrade to version 2.02</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.03-1" date="1571140802">- Upgrade to version 2.03</changelog>
</package>
<package pkgid="ad78710e199fb6fee405ef39b1882a46a1068e4ab0790e3ce53b0d8706f86316" name="asterisk-voicechanger" arch="src">
  <version epoch="0" ver="0.1.957ab53" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.1.957ab53" date="1769601600">- First release of this package</changelog>
</package>
<package pkgid="2f78d2bf514c3938708c91043f94c0098e5ca14523c75a2327fec3341ee896ee" name="boost-nowide" arch="src">
  <version epoch="0" ver="0" rel="20190813.gitec9672b.el8"/>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 0-20171003.gitec9672b" date="1507118400">- Initial packaging</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 0-20171019.gitec9672b" date="1508414400">- Include documentation files
- Adjust to bring in line with guidelines for a header only package
- Place headers in correct place for epel</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 0-20171026.gitec9672b" date="1508414401">- Fix the directory location on EPEL to align with the boost157 package correctly
- Move the docs to a dedicated subpackage</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0-20171027.gitec9672b" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0-20171028.gitec9672b" date="1531396800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0-20171029.gitec9672b" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.10.4-4" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
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  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1466596800">- gp-Use-initial-exec-tls-for-libunwind-s-recursion-flag.patch [bz#1339710]
- Resolves: bz#1339710
  (initalization of 'recursive' tls variable in libunwind stack capturer occasionally triggers deadlock in ceph)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1507809600">- Rebase to 2.6.1 [bz#1431240]
- Removed libunwind usage [bz#1467203]
- Resolves: bz#1431240 
  (gperftools fails to build on s390x, lacks s390x support)
- Resolves: bz#1467203
  (Please, remove libunwind from the gperftools-libs (and 389-ds-base) requirements)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.2-1" date="1512043200">- gperftools 2.6.2 is out!
-
- Most notable change is recently added support for C++17 over-aligned allocation
- operators contributed by Andrey Semashev. I've extended his implementation to
- have roughly same performance as malloc/new. This release also has native
- support for C11 aligned_alloc.
-
- Rest is mostly bug fixes:
-
- * Jianbo Yang has contributed a fix for potentially severe data raceintroduced
-   by malloc fast-path work in gperftools 2.6. This race could cause occasional
-   violation of total thread cache size constraint. See issue #929 for more
-   details.
- * Correct behavior in out-of-memory condition in fast-path cases was restored.
-   This was another bug introduced by fast-path optimization in gperftools 2.6
-   which caused operator new to silently return NULL instead of doing correct
-   C++ OOM handling (calling new_handler and throwing bad_alloc).
- * Khem Raj has contributed couple build fixes for newer glibcs (ucontext_t vs
-   struct ucontext and loff_t definition)
- * Piotr Sikora has contributed build fix for OSX (not building unwind
-   benchmark). This was issue #910 (thanks to Yuriy Solovyov for reporting it).
- * Dorin Lazăr has contributed fix for compiler warning
- * issue #912 (occasional deadlocking calling getenv too early on windows) was
-   fixed. Thanks to github user shangcangriluo for reporting it.
- * Couple earlier lsan-related commits still causing occasional issues linking
-   on OSX has been reverted. See issue #901.
- * Volodimir Krylov has contributed GetProgramInvocationName for FreeBSD
- * changsu lee has contributed couple minor correctness fixes (missing va_end()
-   and missing free() call in rarely executed Symbolize path)
- * Andrew C. Morrow has contributed some more page heap stats. See issue #935.
- * some cases of built-time warnings from various gcc/clang versions about
-   throw() declarations have been fixes.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.3-1" date="1512820800">- gperftools 2.6.3 is out!
-
- Just two fixes were made in this release:
-
- * Stephan Zuercher has contributed a build fix for some recent XCode versions.
-   See issue #942 for more details.
- * assertion failure on some windows builds introduced by 2.6.2 was fixed.
-   Thanks to github user nkeemik for reporting it and testing fix. See issue
-   #944 for more details.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.7-1" date="1525089600">- gperftools 2.7 is out!
-
- Few people contributed minor, but important fixes since rc.
-
- Changes:
-
- * bug in span stats printing introduced by new scalable page heap change was
-   fixed.
- * Christoph Müllner has contributed couple warnings fixes and initial support
-   for aarch64_ilp32 architecture.
- * Ben Dang contributed documentation fix for heap checker.
- * Fabrice Fontaine contributed fixed for linking benchmarks with
-   --disable-static.
- * Holy Wu has added sized deallocation unit tests.
- * Holy Wu has enabled support of sized deallocation (c++14) on recent MSVC.
- * Holy Wu has fixed MSVC build in WIN32_OVERRIDE_ALLOCATORS mode. This closed
-   issue #716.
- * Holy Wu has contributed cleanup of config.h used on windows.
- * Mao Huang has contributed couple simple tcmalloc changes from chromium code
-   base. Making our tcmalloc forks a tiny bit closer.
- * issue #946 that caused compilation failures on some Linux clang installations
-   has been fixed. Much thanks to github user htuch for helping to diagnose
-   issue and proposing a fix.
- * Tulio Magno Quites Machado Filho has contributed build-time fix for PPC (for
-   problem introduced in one of commits since RC).</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8-1" date="1594036800">- gperftools 2.8 is out!
-
- Here are notable changes:
-
- * ProfilerGetStackTrace is now officially supported API for libprofiler.
-   Contributed by Kirill Müller.
- * Build failures on mingw were fixed. This fixed issue #1108.
- * Build failure of page_heap_test on MSVC was fixed.
- * Ryan Macnak contributed fix for compiling linux syscall support on i386 and
-   recent GCCs. This fixed issue #1076.
- * test failures caused by new gcc 10 optimizations were fixed. Same change also
-   fixed tests on clang.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8.1-1" date="1608552000">- gperftools-2.8.1 is out!
-
- Here are notable changes:
-
- * previous release contained change to release memory without page heap lock,
-   but this change had at least one bug that caused to crashes and corruption
-   when running under aggressive decommit mode (this is not default). While we
-   check for other bugs, this feature was reverted. See github issue #1204 and
-   issue #1227.
- * stack traces depth captured by gperftools is now up to 254 levels deep.
-   Thanks to Kerrick Staley for this small but useful tweak.
- * Levon Ter-Grigoryan has contributed small fix for compiler warning.
- * Grant Henke has contributed updated detection of program counter register for
-   OS X on arm64.
- * Tim Gates has contributed small typo fix.
- * Steve Langasek has contributed basic build fixes for riscv64.
- * Isaac Hier and okhowang have contributed premiliminary port of build
-   infrastructure to cmake. This works, but it is very premiliminary.
-   Autotools-based build is the only officially supported build for now.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9-1" date="1613908800">- gperftools 2.9 is out!
-
- Here are notable changes:
-
- * Jarno Rajahalme has contributed fix crashing bug in syscalls support for
-   aarch64.
- * User SSE4 has contributed basic support for Elbrus 2000 architecture (!)
- * Venkatesh Srinivas has contributed cleanup to atomic ops.
- * Đoàn Trần Công Danh has fixed cpu profiler compilation on musl.
- * there is now better backtracing support for aarch64 and riscv. x86-64 with
-   frame pointers now also defaults to this new "generic" frame pointer
-   backtracer.
- * emergency malloc is now enabled by default. Fixes hang on musl when libgcc
-   backtracer is enabled.
- * bunch of legacy config tests has been removed
- * Venkatesh Srinivas has contributed thread-safety annotations support.
- * couple more unit test bugs that caused tcmalloc_unittest to fail on recent
-   clang has been fixed.
- * usage of unsupportable linux_syscall_support.h has been removed from few
-   places. Building with --disable-heap-checker now completely avoids it. Expect
-   complete death of this header in next major release.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9.1-1" date="1614772800">- gperftools 2.9.1 is out!
-
- Minor fixes landed since previous release:
-
- * OSX builds now prefer backtrace() and have somewhat working heap sampling.
- * Incorrect assertion failure was fixed that crashed tcmalloc if assertions
-   were on and sized delete was used. More details in github issue #1254.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.10-1" date="1653912000">- gperftools 2.10 is out!
-
- Here are notable changes:
-
- * Matt T. Proud contributed documentation fix to call Go programming language
-   by it's true name instead of golang.
- * Robert Scott contributed debugallocator feature to use readable (PROT_READ)
-   fence pages. This is activated by TCMALLOC_PAGE_FENCE_READABLE environment
-   veriable.
- * User stdpain contributed fix for cmake detection of libunwind.
- * Natale Patriciello contributed fix for OSX Monterey support.
- * Volodymyr Nikolaichuk contributed support for returning memory back to OS by
-   using mmap with MAP_FIXED and PROT_NONE. It is off by default and enabled by
-   preprocessor define: FREE_MMAP_PROT_NONE. This should help OSes that don't
-   support Linux-style madvise MADV_DONTNEED or BSD-style MADV_FREE.
- * Jingyun Hua has contributed basic support for LoongArch.
- * Github issue #1338 of failing to build on some recent musl versions has been
-   fixed.
- * Github issue #1321 of failing to ship cmake bits with .tar.gz archive has
-   been fixed.</changelog>
</package>
<package pkgid="2b127749bccec180997ef8dc0e6304366dcf1e2ff3bba27fe7f1fc80d1330fce" name="gperftools" arch="src">
  <version epoch="0" ver="2.11" rel="1.el8"/>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1507809600">- Rebase to 2.6.1 [bz#1431240]
- Removed libunwind usage [bz#1467203]
- Resolves: bz#1431240 
  (gperftools fails to build on s390x, lacks s390x support)
- Resolves: bz#1467203
  (Please, remove libunwind from the gperftools-libs (and 389-ds-base) requirements)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.2-1" date="1512043200">- gperftools 2.6.2 is out!
-
- Most notable change is recently added support for C++17 over-aligned allocation
- operators contributed by Andrey Semashev. I've extended his implementation to
- have roughly same performance as malloc/new. This release also has native
- support for C11 aligned_alloc.
-
- Rest is mostly bug fixes:
-
- * Jianbo Yang has contributed a fix for potentially severe data raceintroduced
-   by malloc fast-path work in gperftools 2.6. This race could cause occasional
-   violation of total thread cache size constraint. See issue #929 for more
-   details.
- * Correct behavior in out-of-memory condition in fast-path cases was restored.
-   This was another bug introduced by fast-path optimization in gperftools 2.6
-   which caused operator new to silently return NULL instead of doing correct
-   C++ OOM handling (calling new_handler and throwing bad_alloc).
- * Khem Raj has contributed couple build fixes for newer glibcs (ucontext_t vs
-   struct ucontext and loff_t definition)
- * Piotr Sikora has contributed build fix for OSX (not building unwind
-   benchmark). This was issue #910 (thanks to Yuriy Solovyov for reporting it).
- * Dorin Lazăr has contributed fix for compiler warning
- * issue #912 (occasional deadlocking calling getenv too early on windows) was
-   fixed. Thanks to github user shangcangriluo for reporting it.
- * Couple earlier lsan-related commits still causing occasional issues linking
-   on OSX has been reverted. See issue #901.
- * Volodimir Krylov has contributed GetProgramInvocationName for FreeBSD
- * changsu lee has contributed couple minor correctness fixes (missing va_end()
-   and missing free() call in rarely executed Symbolize path)
- * Andrew C. Morrow has contributed some more page heap stats. See issue #935.
- * some cases of built-time warnings from various gcc/clang versions about
-   throw() declarations have been fixes.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.3-1" date="1512820800">- gperftools 2.6.3 is out!
-
- Just two fixes were made in this release:
-
- * Stephan Zuercher has contributed a build fix for some recent XCode versions.
-   See issue #942 for more details.
- * assertion failure on some windows builds introduced by 2.6.2 was fixed.
-   Thanks to github user nkeemik for reporting it and testing fix. See issue
-   #944 for more details.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.7-1" date="1525089600">- gperftools 2.7 is out!
-
- Few people contributed minor, but important fixes since rc.
-
- Changes:
-
- * bug in span stats printing introduced by new scalable page heap change was
-   fixed.
- * Christoph Müllner has contributed couple warnings fixes and initial support
-   for aarch64_ilp32 architecture.
- * Ben Dang contributed documentation fix for heap checker.
- * Fabrice Fontaine contributed fixed for linking benchmarks with
-   --disable-static.
- * Holy Wu has added sized deallocation unit tests.
- * Holy Wu has enabled support of sized deallocation (c++14) on recent MSVC.
- * Holy Wu has fixed MSVC build in WIN32_OVERRIDE_ALLOCATORS mode. This closed
-   issue #716.
- * Holy Wu has contributed cleanup of config.h used on windows.
- * Mao Huang has contributed couple simple tcmalloc changes from chromium code
-   base. Making our tcmalloc forks a tiny bit closer.
- * issue #946 that caused compilation failures on some Linux clang installations
-   has been fixed. Much thanks to github user htuch for helping to diagnose
-   issue and proposing a fix.
- * Tulio Magno Quites Machado Filho has contributed build-time fix for PPC (for
-   problem introduced in one of commits since RC).</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8-1" date="1594036800">- gperftools 2.8 is out!
-
- Here are notable changes:
-
- * ProfilerGetStackTrace is now officially supported API for libprofiler.
-   Contributed by Kirill Müller.
- * Build failures on mingw were fixed. This fixed issue #1108.
- * Build failure of page_heap_test on MSVC was fixed.
- * Ryan Macnak contributed fix for compiling linux syscall support on i386 and
-   recent GCCs. This fixed issue #1076.
- * test failures caused by new gcc 10 optimizations were fixed. Same change also
-   fixed tests on clang.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8.1-1" date="1608552000">- gperftools-2.8.1 is out!
-
- Here are notable changes:
-
- * previous release contained change to release memory without page heap lock,
-   but this change had at least one bug that caused to crashes and corruption
-   when running under aggressive decommit mode (this is not default). While we
-   check for other bugs, this feature was reverted. See github issue #1204 and
-   issue #1227.
- * stack traces depth captured by gperftools is now up to 254 levels deep.
-   Thanks to Kerrick Staley for this small but useful tweak.
- * Levon Ter-Grigoryan has contributed small fix for compiler warning.
- * Grant Henke has contributed updated detection of program counter register for
-   OS X on arm64.
- * Tim Gates has contributed small typo fix.
- * Steve Langasek has contributed basic build fixes for riscv64.
- * Isaac Hier and okhowang have contributed premiliminary port of build
-   infrastructure to cmake. This works, but it is very premiliminary.
-   Autotools-based build is the only officially supported build for now.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9-1" date="1613908800">- gperftools 2.9 is out!
-
- Here are notable changes:
-
- * Jarno Rajahalme has contributed fix crashing bug in syscalls support for
-   aarch64.
- * User SSE4 has contributed basic support for Elbrus 2000 architecture (!)
- * Venkatesh Srinivas has contributed cleanup to atomic ops.
- * Đoàn Trần Công Danh has fixed cpu profiler compilation on musl.
- * there is now better backtracing support for aarch64 and riscv. x86-64 with
-   frame pointers now also defaults to this new "generic" frame pointer
-   backtracer.
- * emergency malloc is now enabled by default. Fixes hang on musl when libgcc
-   backtracer is enabled.
- * bunch of legacy config tests has been removed
- * Venkatesh Srinivas has contributed thread-safety annotations support.
- * couple more unit test bugs that caused tcmalloc_unittest to fail on recent
-   clang has been fixed.
- * usage of unsupportable linux_syscall_support.h has been removed from few
-   places. Building with --disable-heap-checker now completely avoids it. Expect
-   complete death of this header in next major release.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9.1-1" date="1614772800">- gperftools 2.9.1 is out!
-
- Minor fixes landed since previous release:
-
- * OSX builds now prefer backtrace() and have somewhat working heap sampling.
- * Incorrect assertion failure was fixed that crashed tcmalloc if assertions
-   were on and sized delete was used. More details in github issue #1254.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.10-1" date="1653912000">- gperftools 2.10 is out!
-
- Here are notable changes:
-
- * Matt T. Proud contributed documentation fix to call Go programming language
-   by it's true name instead of golang.
- * Robert Scott contributed debugallocator feature to use readable (PROT_READ)
-   fence pages. This is activated by TCMALLOC_PAGE_FENCE_READABLE environment
-   veriable.
- * User stdpain contributed fix for cmake detection of libunwind.
- * Natale Patriciello contributed fix for OSX Monterey support.
- * Volodymyr Nikolaichuk contributed support for returning memory back to OS by
-   using mmap with MAP_FIXED and PROT_NONE. It is off by default and enabled by
-   preprocessor define: FREE_MMAP_PROT_NONE. This should help OSes that don't
-   support Linux-style madvise MADV_DONTNEED or BSD-style MADV_FREE.
- * Jingyun Hua has contributed basic support for LoongArch.
- * Github issue #1338 of failing to build on some recent musl versions has been
-   fixed.
- * Github issue #1321 of failing to ship cmake bits with .tar.gz archive has
-   been fixed.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.11-1" date="1692014400">- gperftools 2.11 is out!
-
- Most notable change is that Linux/aarch64 and Linux/riscv are now fully
- supported. That is, all unit tests pass on those architectures (previously the
- heap leak checker was broken).
-
- Also notable is that heap leak checker support is officially deprecated as of
- this release. All bug fixes from now are on a best effort basis. For clarity we
- also declare that it is only expected to work (for some definition of work) on
- Linux/x86 (all kinds), Linux/aarch64, Linux/arm, Linux/ppc (untested as of this
- writing) and Linux/mips (untested as well). While some functionality worked in
- the past on BSDs, it was never fully functional; and will never be. We strongly
- recommend everyone to switch to asan and friends.
-
- For major internal changes it is also worth mentioning that we now fully
- switched to C++-11 std::atomic. All custom OS- and arch-specific atomic bits
- have been removed at last.
-
- Another notable change is that mmap and sbrk hooks facility is now no-op. We
- keep API and ABI for formal compatibility, but the calls to add mmap/sbrk hooks
- do nothing and return an error (whenever possible as part of API). There seem
- to be no users of it anyways, and mmap replacement API that is part of that
- facility really screwed up 64-bit offsets on (some/most) 32-bit systems.
- Internally for heap profiler and heap checker we have a new, but non-public API
- (see mmap_hook.h).
-
- Most tests now pass on NetBSD x86-64 (I tested on version 9.2). And only one
- that fails is new stacktrace test for stacktraces from signal handler (so there
- could be some imperfections for cpu profiles).
-
- We don't warn people away from the libgcc stacktrace capturing method anymore.
- In fact users on most recent glibc-s are advised to use it (pass
- --enable-libgcc-unwinder-by-default). This is thanks to the dl_find_object API
- offered by glibc which allows this implementation to be fully
- async-signal-safe. Modern Linux distros should from now on build their
- gperftools package with this enabled (other than those built on top of musl).
-
- generic_fp and generic_fp_unsafe stacktrace capturing methods have been
- expanded for more architectures and even some basic non-Linux support. We have
- completely removed old x86-specific frame pointer stacktrace implementation in
- favor of those 2. _unsafe one should be roughly equivalent to the old x86
- method. And 'safe' one is recommended as a new default for those who want
- FP-based stacktracing. Safe implementation robustly checks memory before
- accessing it, preventing unlikely, but not impossible crashes when frame
- pointers are bogus.
-
- On platforms that support it, we now build gperftools with
- "-fno-omit-frame-pointer -momit-leaf-frame-pointer". This makes gperftools
- mostly frame-pointer-ful, but without performance hit in places that matter
- (this is how Google builds their binaries BTW). That should cover gcc (at
- least) on x86, aarch64 and riscv. Intention for this change is to make
- distro-shipped libtcmalloc.so compatible with frame-pointer stacktrace
- capturing (for those who still do heap profiling, for example). Of course,
- passing --enable-frame-pointers still gives you full frame pointers (i.e. even
- for leaf functions).
-
- There is now support for detecting actual page size at runtime. tcmalloc will
- now allocate memory in units of this page size. It particularly helps on arms
- with 64k pages to return memory back to the kernel. But it is somewhat
- controversial, because it effectively bumps tcmalloc logical page size on those
- machines potentially increasing fragmentation. In any case, there is now a new
- environment variable TCMALLOC_OVERRIDE_PAGESIZE allowing people to override
- this check. I.e. to either reduce effective page size down to tcmalloc's
- logical page size or to increase it.
-
- MallocExtension::MarkThreadTemporarilyIdle has been changed to be identical to
- MarkThreadIdle. MarkThreadTemporarilyIdle is believed to be unused, anyways.
- See issue #880 for details.
-
- There are a whole bunch of smaller fixes. Many of those smaller fixes had no
- associated ticket, but some had. People are advised to see here for list of
- notable tickets closed in this release:
- https://github.com/gperftools/gperftools/issues?q=label%3Afixed-in-2.11+
-
- Some of those tickets are quite notable (fixes for rare deadlocks in cpu
- profiler ProfilerStop or while capturing heap growth stacktraces (aka
- growthz)).
-
- Here is list of notable contributions:
-
- * Chris Cambly has contributed initial support for AIX
- * Ali Saidi has contributed SpinlockPause implementation for aarch64
- * Henrik Reinstädtler has contributed fix for cpuprofiler on aarch64 OSX
- * Gabriel Marin has backported Chromium's commit for always sanity checking
-   large frees
- * User zhangyiru has contributed a fix to report the number of leaked bytes as
-   size_t instead of (usually 32-bit) int.
- * Sergey Fedorov has contributed some fix for building on older ppc-based
-   OSX-es
- * User tigeran has removed unused using declaration
- * Artem Polyakov has contributed auto-detection of several MPI systems w.r.t.
-   filenames used by HEAPPROFILE and CPUPROFILE environment variables. Also, we
-   now support HEAPPROFILE_USE_PID and CPUPROFILE_USE_PID environment variables
-   that force profile filenames to have pid appended. Which will be useful for
-   some programs that fork for parallelism. See #1263 for details.
- * Ken Raffenetti has extended MPI detection mentioned above with detection of
-   MPICH system.
-
- Thanks a lot!</changelog>
</package>
<package pkgid="876060a24ae5870e36fb807bca673b92794010a0a25baf90b03c31ebe2304261" name="gperftools" arch="src">
  <version epoch="0" ver="2.12" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.2-1" date="1512043200">- gperftools 2.6.2 is out!
-
- Most notable change is recently added support for C++17 over-aligned allocation
- operators contributed by Andrey Semashev. I've extended his implementation to
- have roughly same performance as malloc/new. This release also has native
- support for C11 aligned_alloc.
-
- Rest is mostly bug fixes:
-
- * Jianbo Yang has contributed a fix for potentially severe data raceintroduced
-   by malloc fast-path work in gperftools 2.6. This race could cause occasional
-   violation of total thread cache size constraint. See issue #929 for more
-   details.
- * Correct behavior in out-of-memory condition in fast-path cases was restored.
-   This was another bug introduced by fast-path optimization in gperftools 2.6
-   which caused operator new to silently return NULL instead of doing correct
-   C++ OOM handling (calling new_handler and throwing bad_alloc).
- * Khem Raj has contributed couple build fixes for newer glibcs (ucontext_t vs
-   struct ucontext and loff_t definition)
- * Piotr Sikora has contributed build fix for OSX (not building unwind
-   benchmark). This was issue #910 (thanks to Yuriy Solovyov for reporting it).
- * Dorin Lazăr has contributed fix for compiler warning
- * issue #912 (occasional deadlocking calling getenv too early on windows) was
-   fixed. Thanks to github user shangcangriluo for reporting it.
- * Couple earlier lsan-related commits still causing occasional issues linking
-   on OSX has been reverted. See issue #901.
- * Volodimir Krylov has contributed GetProgramInvocationName for FreeBSD
- * changsu lee has contributed couple minor correctness fixes (missing va_end()
-   and missing free() call in rarely executed Symbolize path)
- * Andrew C. Morrow has contributed some more page heap stats. See issue #935.
- * some cases of built-time warnings from various gcc/clang versions about
-   throw() declarations have been fixes.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.3-1" date="1512820800">- gperftools 2.6.3 is out!
-
- Just two fixes were made in this release:
-
- * Stephan Zuercher has contributed a build fix for some recent XCode versions.
-   See issue #942 for more details.
- * assertion failure on some windows builds introduced by 2.6.2 was fixed.
-   Thanks to github user nkeemik for reporting it and testing fix. See issue
-   #944 for more details.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.7-1" date="1525089600">- gperftools 2.7 is out!
-
- Few people contributed minor, but important fixes since rc.
-
- Changes:
-
- * bug in span stats printing introduced by new scalable page heap change was
-   fixed.
- * Christoph Müllner has contributed couple warnings fixes and initial support
-   for aarch64_ilp32 architecture.
- * Ben Dang contributed documentation fix for heap checker.
- * Fabrice Fontaine contributed fixed for linking benchmarks with
-   --disable-static.
- * Holy Wu has added sized deallocation unit tests.
- * Holy Wu has enabled support of sized deallocation (c++14) on recent MSVC.
- * Holy Wu has fixed MSVC build in WIN32_OVERRIDE_ALLOCATORS mode. This closed
-   issue #716.
- * Holy Wu has contributed cleanup of config.h used on windows.
- * Mao Huang has contributed couple simple tcmalloc changes from chromium code
-   base. Making our tcmalloc forks a tiny bit closer.
- * issue #946 that caused compilation failures on some Linux clang installations
-   has been fixed. Much thanks to github user htuch for helping to diagnose
-   issue and proposing a fix.
- * Tulio Magno Quites Machado Filho has contributed build-time fix for PPC (for
-   problem introduced in one of commits since RC).</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8-1" date="1594036800">- gperftools 2.8 is out!
-
- Here are notable changes:
-
- * ProfilerGetStackTrace is now officially supported API for libprofiler.
-   Contributed by Kirill Müller.
- * Build failures on mingw were fixed. This fixed issue #1108.
- * Build failure of page_heap_test on MSVC was fixed.
- * Ryan Macnak contributed fix for compiling linux syscall support on i386 and
-   recent GCCs. This fixed issue #1076.
- * test failures caused by new gcc 10 optimizations were fixed. Same change also
-   fixed tests on clang.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8.1-1" date="1608552000">- gperftools-2.8.1 is out!
-
- Here are notable changes:
-
- * previous release contained change to release memory without page heap lock,
-   but this change had at least one bug that caused to crashes and corruption
-   when running under aggressive decommit mode (this is not default). While we
-   check for other bugs, this feature was reverted. See github issue #1204 and
-   issue #1227.
- * stack traces depth captured by gperftools is now up to 254 levels deep.
-   Thanks to Kerrick Staley for this small but useful tweak.
- * Levon Ter-Grigoryan has contributed small fix for compiler warning.
- * Grant Henke has contributed updated detection of program counter register for
-   OS X on arm64.
- * Tim Gates has contributed small typo fix.
- * Steve Langasek has contributed basic build fixes for riscv64.
- * Isaac Hier and okhowang have contributed premiliminary port of build
-   infrastructure to cmake. This works, but it is very premiliminary.
-   Autotools-based build is the only officially supported build for now.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9-1" date="1613908800">- gperftools 2.9 is out!
-
- Here are notable changes:
-
- * Jarno Rajahalme has contributed fix crashing bug in syscalls support for
-   aarch64.
- * User SSE4 has contributed basic support for Elbrus 2000 architecture (!)
- * Venkatesh Srinivas has contributed cleanup to atomic ops.
- * Đoàn Trần Công Danh has fixed cpu profiler compilation on musl.
- * there is now better backtracing support for aarch64 and riscv. x86-64 with
-   frame pointers now also defaults to this new "generic" frame pointer
-   backtracer.
- * emergency malloc is now enabled by default. Fixes hang on musl when libgcc
-   backtracer is enabled.
- * bunch of legacy config tests has been removed
- * Venkatesh Srinivas has contributed thread-safety annotations support.
- * couple more unit test bugs that caused tcmalloc_unittest to fail on recent
-   clang has been fixed.
- * usage of unsupportable linux_syscall_support.h has been removed from few
-   places. Building with --disable-heap-checker now completely avoids it. Expect
-   complete death of this header in next major release.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9.1-1" date="1614772800">- gperftools 2.9.1 is out!
-
- Minor fixes landed since previous release:
-
- * OSX builds now prefer backtrace() and have somewhat working heap sampling.
- * Incorrect assertion failure was fixed that crashed tcmalloc if assertions
-   were on and sized delete was used. More details in github issue #1254.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.10-1" date="1653912000">- gperftools 2.10 is out!
-
- Here are notable changes:
-
- * Matt T. Proud contributed documentation fix to call Go programming language
-   by it's true name instead of golang.
- * Robert Scott contributed debugallocator feature to use readable (PROT_READ)
-   fence pages. This is activated by TCMALLOC_PAGE_FENCE_READABLE environment
-   veriable.
- * User stdpain contributed fix for cmake detection of libunwind.
- * Natale Patriciello contributed fix for OSX Monterey support.
- * Volodymyr Nikolaichuk contributed support for returning memory back to OS by
-   using mmap with MAP_FIXED and PROT_NONE. It is off by default and enabled by
-   preprocessor define: FREE_MMAP_PROT_NONE. This should help OSes that don't
-   support Linux-style madvise MADV_DONTNEED or BSD-style MADV_FREE.
- * Jingyun Hua has contributed basic support for LoongArch.
- * Github issue #1338 of failing to build on some recent musl versions has been
-   fixed.
- * Github issue #1321 of failing to ship cmake bits with .tar.gz archive has
-   been fixed.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.11-1" date="1692014400">- gperftools 2.11 is out!
-
- Most notable change is that Linux/aarch64 and Linux/riscv are now fully
- supported. That is, all unit tests pass on those architectures (previously the
- heap leak checker was broken).
-
- Also notable is that heap leak checker support is officially deprecated as of
- this release. All bug fixes from now are on a best effort basis. For clarity we
- also declare that it is only expected to work (for some definition of work) on
- Linux/x86 (all kinds), Linux/aarch64, Linux/arm, Linux/ppc (untested as of this
- writing) and Linux/mips (untested as well). While some functionality worked in
- the past on BSDs, it was never fully functional; and will never be. We strongly
- recommend everyone to switch to asan and friends.
-
- For major internal changes it is also worth mentioning that we now fully
- switched to C++-11 std::atomic. All custom OS- and arch-specific atomic bits
- have been removed at last.
-
- Another notable change is that mmap and sbrk hooks facility is now no-op. We
- keep API and ABI for formal compatibility, but the calls to add mmap/sbrk hooks
- do nothing and return an error (whenever possible as part of API). There seem
- to be no users of it anyways, and mmap replacement API that is part of that
- facility really screwed up 64-bit offsets on (some/most) 32-bit systems.
- Internally for heap profiler and heap checker we have a new, but non-public API
- (see mmap_hook.h).
-
- Most tests now pass on NetBSD x86-64 (I tested on version 9.2). And only one
- that fails is new stacktrace test for stacktraces from signal handler (so there
- could be some imperfections for cpu profiles).
-
- We don't warn people away from the libgcc stacktrace capturing method anymore.
- In fact users on most recent glibc-s are advised to use it (pass
- --enable-libgcc-unwinder-by-default). This is thanks to the dl_find_object API
- offered by glibc which allows this implementation to be fully
- async-signal-safe. Modern Linux distros should from now on build their
- gperftools package with this enabled (other than those built on top of musl).
-
- generic_fp and generic_fp_unsafe stacktrace capturing methods have been
- expanded for more architectures and even some basic non-Linux support. We have
- completely removed old x86-specific frame pointer stacktrace implementation in
- favor of those 2. _unsafe one should be roughly equivalent to the old x86
- method. And 'safe' one is recommended as a new default for those who want
- FP-based stacktracing. Safe implementation robustly checks memory before
- accessing it, preventing unlikely, but not impossible crashes when frame
- pointers are bogus.
-
- On platforms that support it, we now build gperftools with
- "-fno-omit-frame-pointer -momit-leaf-frame-pointer". This makes gperftools
- mostly frame-pointer-ful, but without performance hit in places that matter
- (this is how Google builds their binaries BTW). That should cover gcc (at
- least) on x86, aarch64 and riscv. Intention for this change is to make
- distro-shipped libtcmalloc.so compatible with frame-pointer stacktrace
- capturing (for those who still do heap profiling, for example). Of course,
- passing --enable-frame-pointers still gives you full frame pointers (i.e. even
- for leaf functions).
-
- There is now support for detecting actual page size at runtime. tcmalloc will
- now allocate memory in units of this page size. It particularly helps on arms
- with 64k pages to return memory back to the kernel. But it is somewhat
- controversial, because it effectively bumps tcmalloc logical page size on those
- machines potentially increasing fragmentation. In any case, there is now a new
- environment variable TCMALLOC_OVERRIDE_PAGESIZE allowing people to override
- this check. I.e. to either reduce effective page size down to tcmalloc's
- logical page size or to increase it.
-
- MallocExtension::MarkThreadTemporarilyIdle has been changed to be identical to
- MarkThreadIdle. MarkThreadTemporarilyIdle is believed to be unused, anyways.
- See issue #880 for details.
-
- There are a whole bunch of smaller fixes. Many of those smaller fixes had no
- associated ticket, but some had. People are advised to see here for list of
- notable tickets closed in this release:
- https://github.com/gperftools/gperftools/issues?q=label%3Afixed-in-2.11+
-
- Some of those tickets are quite notable (fixes for rare deadlocks in cpu
- profiler ProfilerStop or while capturing heap growth stacktraces (aka
- growthz)).
-
- Here is list of notable contributions:
-
- * Chris Cambly has contributed initial support for AIX
- * Ali Saidi has contributed SpinlockPause implementation for aarch64
- * Henrik Reinstädtler has contributed fix for cpuprofiler on aarch64 OSX
- * Gabriel Marin has backported Chromium's commit for always sanity checking
-   large frees
- * User zhangyiru has contributed a fix to report the number of leaked bytes as
-   size_t instead of (usually 32-bit) int.
- * Sergey Fedorov has contributed some fix for building on older ppc-based
-   OSX-es
- * User tigeran has removed unused using declaration
- * Artem Polyakov has contributed auto-detection of several MPI systems w.r.t.
-   filenames used by HEAPPROFILE and CPUPROFILE environment variables. Also, we
-   now support HEAPPROFILE_USE_PID and CPUPROFILE_USE_PID environment variables
-   that force profile filenames to have pid appended. Which will be useful for
-   some programs that fork for parallelism. See #1263 for details.
- * Ken Raffenetti has extended MPI detection mentioned above with detection of
-   MPICH system.
-
- Thanks a lot!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.12-1" date="1692878400">- Brett T. Warden contributed one significant fix. After a change in the previous
- release, we installed broken pkg-config files. Brett noticed and fixed that.
- Huge thanks!</changelog>
</package>
<package pkgid="bf542015b6528e649dfe0ca0fd850adbe3354501c36c15fd2d708a013185225a" name="gperftools" arch="src">
  <version epoch="0" ver="2.13" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.3-1" date="1512820800">- gperftools 2.6.3 is out!
-
- Just two fixes were made in this release:
-
- * Stephan Zuercher has contributed a build fix for some recent XCode versions.
-   See issue #942 for more details.
- * assertion failure on some windows builds introduced by 2.6.2 was fixed.
-   Thanks to github user nkeemik for reporting it and testing fix. See issue
-   #944 for more details.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.7-1" date="1525089600">- gperftools 2.7 is out!
-
- Few people contributed minor, but important fixes since rc.
-
- Changes:
-
- * bug in span stats printing introduced by new scalable page heap change was
-   fixed.
- * Christoph Müllner has contributed couple warnings fixes and initial support
-   for aarch64_ilp32 architecture.
- * Ben Dang contributed documentation fix for heap checker.
- * Fabrice Fontaine contributed fixed for linking benchmarks with
-   --disable-static.
- * Holy Wu has added sized deallocation unit tests.
- * Holy Wu has enabled support of sized deallocation (c++14) on recent MSVC.
- * Holy Wu has fixed MSVC build in WIN32_OVERRIDE_ALLOCATORS mode. This closed
-   issue #716.
- * Holy Wu has contributed cleanup of config.h used on windows.
- * Mao Huang has contributed couple simple tcmalloc changes from chromium code
-   base. Making our tcmalloc forks a tiny bit closer.
- * issue #946 that caused compilation failures on some Linux clang installations
-   has been fixed. Much thanks to github user htuch for helping to diagnose
-   issue and proposing a fix.
- * Tulio Magno Quites Machado Filho has contributed build-time fix for PPC (for
-   problem introduced in one of commits since RC).</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8-1" date="1594036800">- gperftools 2.8 is out!
-
- Here are notable changes:
-
- * ProfilerGetStackTrace is now officially supported API for libprofiler.
-   Contributed by Kirill Müller.
- * Build failures on mingw were fixed. This fixed issue #1108.
- * Build failure of page_heap_test on MSVC was fixed.
- * Ryan Macnak contributed fix for compiling linux syscall support on i386 and
-   recent GCCs. This fixed issue #1076.
- * test failures caused by new gcc 10 optimizations were fixed. Same change also
-   fixed tests on clang.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8.1-1" date="1608552000">- gperftools-2.8.1 is out!
-
- Here are notable changes:
-
- * previous release contained change to release memory without page heap lock,
-   but this change had at least one bug that caused to crashes and corruption
-   when running under aggressive decommit mode (this is not default). While we
-   check for other bugs, this feature was reverted. See github issue #1204 and
-   issue #1227.
- * stack traces depth captured by gperftools is now up to 254 levels deep.
-   Thanks to Kerrick Staley for this small but useful tweak.
- * Levon Ter-Grigoryan has contributed small fix for compiler warning.
- * Grant Henke has contributed updated detection of program counter register for
-   OS X on arm64.
- * Tim Gates has contributed small typo fix.
- * Steve Langasek has contributed basic build fixes for riscv64.
- * Isaac Hier and okhowang have contributed premiliminary port of build
-   infrastructure to cmake. This works, but it is very premiliminary.
-   Autotools-based build is the only officially supported build for now.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9-1" date="1613908800">- gperftools 2.9 is out!
-
- Here are notable changes:
-
- * Jarno Rajahalme has contributed fix crashing bug in syscalls support for
-   aarch64.
- * User SSE4 has contributed basic support for Elbrus 2000 architecture (!)
- * Venkatesh Srinivas has contributed cleanup to atomic ops.
- * Đoàn Trần Công Danh has fixed cpu profiler compilation on musl.
- * there is now better backtracing support for aarch64 and riscv. x86-64 with
-   frame pointers now also defaults to this new "generic" frame pointer
-   backtracer.
- * emergency malloc is now enabled by default. Fixes hang on musl when libgcc
-   backtracer is enabled.
- * bunch of legacy config tests has been removed
- * Venkatesh Srinivas has contributed thread-safety annotations support.
- * couple more unit test bugs that caused tcmalloc_unittest to fail on recent
-   clang has been fixed.
- * usage of unsupportable linux_syscall_support.h has been removed from few
-   places. Building with --disable-heap-checker now completely avoids it. Expect
-   complete death of this header in next major release.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9.1-1" date="1614772800">- gperftools 2.9.1 is out!
-
- Minor fixes landed since previous release:
-
- * OSX builds now prefer backtrace() and have somewhat working heap sampling.
- * Incorrect assertion failure was fixed that crashed tcmalloc if assertions
-   were on and sized delete was used. More details in github issue #1254.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.10-1" date="1653912000">- gperftools 2.10 is out!
-
- Here are notable changes:
-
- * Matt T. Proud contributed documentation fix to call Go programming language
-   by it's true name instead of golang.
- * Robert Scott contributed debugallocator feature to use readable (PROT_READ)
-   fence pages. This is activated by TCMALLOC_PAGE_FENCE_READABLE environment
-   veriable.
- * User stdpain contributed fix for cmake detection of libunwind.
- * Natale Patriciello contributed fix for OSX Monterey support.
- * Volodymyr Nikolaichuk contributed support for returning memory back to OS by
-   using mmap with MAP_FIXED and PROT_NONE. It is off by default and enabled by
-   preprocessor define: FREE_MMAP_PROT_NONE. This should help OSes that don't
-   support Linux-style madvise MADV_DONTNEED or BSD-style MADV_FREE.
- * Jingyun Hua has contributed basic support for LoongArch.
- * Github issue #1338 of failing to build on some recent musl versions has been
-   fixed.
- * Github issue #1321 of failing to ship cmake bits with .tar.gz archive has
-   been fixed.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.11-1" date="1692014400">- gperftools 2.11 is out!
-
- Most notable change is that Linux/aarch64 and Linux/riscv are now fully
- supported. That is, all unit tests pass on those architectures (previously the
- heap leak checker was broken).
-
- Also notable is that heap leak checker support is officially deprecated as of
- this release. All bug fixes from now are on a best effort basis. For clarity we
- also declare that it is only expected to work (for some definition of work) on
- Linux/x86 (all kinds), Linux/aarch64, Linux/arm, Linux/ppc (untested as of this
- writing) and Linux/mips (untested as well). While some functionality worked in
- the past on BSDs, it was never fully functional; and will never be. We strongly
- recommend everyone to switch to asan and friends.
-
- For major internal changes it is also worth mentioning that we now fully
- switched to C++-11 std::atomic. All custom OS- and arch-specific atomic bits
- have been removed at last.
-
- Another notable change is that mmap and sbrk hooks facility is now no-op. We
- keep API and ABI for formal compatibility, but the calls to add mmap/sbrk hooks
- do nothing and return an error (whenever possible as part of API). There seem
- to be no users of it anyways, and mmap replacement API that is part of that
- facility really screwed up 64-bit offsets on (some/most) 32-bit systems.
- Internally for heap profiler and heap checker we have a new, but non-public API
- (see mmap_hook.h).
-
- Most tests now pass on NetBSD x86-64 (I tested on version 9.2). And only one
- that fails is new stacktrace test for stacktraces from signal handler (so there
- could be some imperfections for cpu profiles).
-
- We don't warn people away from the libgcc stacktrace capturing method anymore.
- In fact users on most recent glibc-s are advised to use it (pass
- --enable-libgcc-unwinder-by-default). This is thanks to the dl_find_object API
- offered by glibc which allows this implementation to be fully
- async-signal-safe. Modern Linux distros should from now on build their
- gperftools package with this enabled (other than those built on top of musl).
-
- generic_fp and generic_fp_unsafe stacktrace capturing methods have been
- expanded for more architectures and even some basic non-Linux support. We have
- completely removed old x86-specific frame pointer stacktrace implementation in
- favor of those 2. _unsafe one should be roughly equivalent to the old x86
- method. And 'safe' one is recommended as a new default for those who want
- FP-based stacktracing. Safe implementation robustly checks memory before
- accessing it, preventing unlikely, but not impossible crashes when frame
- pointers are bogus.
-
- On platforms that support it, we now build gperftools with
- "-fno-omit-frame-pointer -momit-leaf-frame-pointer". This makes gperftools
- mostly frame-pointer-ful, but without performance hit in places that matter
- (this is how Google builds their binaries BTW). That should cover gcc (at
- least) on x86, aarch64 and riscv. Intention for this change is to make
- distro-shipped libtcmalloc.so compatible with frame-pointer stacktrace
- capturing (for those who still do heap profiling, for example). Of course,
- passing --enable-frame-pointers still gives you full frame pointers (i.e. even
- for leaf functions).
-
- There is now support for detecting actual page size at runtime. tcmalloc will
- now allocate memory in units of this page size. It particularly helps on arms
- with 64k pages to return memory back to the kernel. But it is somewhat
- controversial, because it effectively bumps tcmalloc logical page size on those
- machines potentially increasing fragmentation. In any case, there is now a new
- environment variable TCMALLOC_OVERRIDE_PAGESIZE allowing people to override
- this check. I.e. to either reduce effective page size down to tcmalloc's
- logical page size or to increase it.
-
- MallocExtension::MarkThreadTemporarilyIdle has been changed to be identical to
- MarkThreadIdle. MarkThreadTemporarilyIdle is believed to be unused, anyways.
- See issue #880 for details.
-
- There are a whole bunch of smaller fixes. Many of those smaller fixes had no
- associated ticket, but some had. People are advised to see here for list of
- notable tickets closed in this release:
- https://github.com/gperftools/gperftools/issues?q=label%3Afixed-in-2.11+
-
- Some of those tickets are quite notable (fixes for rare deadlocks in cpu
- profiler ProfilerStop or while capturing heap growth stacktraces (aka
- growthz)).
-
- Here is list of notable contributions:
-
- * Chris Cambly has contributed initial support for AIX
- * Ali Saidi has contributed SpinlockPause implementation for aarch64
- * Henrik Reinstädtler has contributed fix for cpuprofiler on aarch64 OSX
- * Gabriel Marin has backported Chromium's commit for always sanity checking
-   large frees
- * User zhangyiru has contributed a fix to report the number of leaked bytes as
-   size_t instead of (usually 32-bit) int.
- * Sergey Fedorov has contributed some fix for building on older ppc-based
-   OSX-es
- * User tigeran has removed unused using declaration
- * Artem Polyakov has contributed auto-detection of several MPI systems w.r.t.
-   filenames used by HEAPPROFILE and CPUPROFILE environment variables. Also, we
-   now support HEAPPROFILE_USE_PID and CPUPROFILE_USE_PID environment variables
-   that force profile filenames to have pid appended. Which will be useful for
-   some programs that fork for parallelism. See #1263 for details.
- * Ken Raffenetti has extended MPI detection mentioned above with detection of
-   MPICH system.
-
- Thanks a lot!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.12-1" date="1692878400">- Brett T. Warden contributed one significant fix. After a change in the previous
- release, we installed broken pkg-config files. Brett noticed and fixed that.
- Huge thanks!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.13-1" date="1694433600">- gperftools 2.13 is out!
-
- This release includes a few minor fixes:
-
- * Ivan Dlugos has fixed some issues with cmake and config.h defines.
- * 32-bit builds no longer require 64-bit atomics (which we wrongly introduced
-   in 2.11 and which broke builds on some 32-bit architectures).
- * generic_fp backtracing method now uses robust address probing method. The
-   previous approach had occasional false positives, which caused occasional
-   rare crashes.
- * In some cases, MSVC generated TrivialOnce machine code that deadlocked
-   programs on startup. The issue is now fixed.</changelog>
</package>
<package pkgid="416bcfe59bf6333817f6f3ae7bab4803eabc8715941f00c6d75c21020aa7504c" name="gperftools" arch="src">
  <version epoch="0" ver="2.14" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.7-1" date="1525089600">- gperftools 2.7 is out!
-
- Few people contributed minor, but important fixes since rc.
-
- Changes:
-
- * bug in span stats printing introduced by new scalable page heap change was
-   fixed.
- * Christoph Müllner has contributed couple warnings fixes and initial support
-   for aarch64_ilp32 architecture.
- * Ben Dang contributed documentation fix for heap checker.
- * Fabrice Fontaine contributed fixed for linking benchmarks with
-   --disable-static.
- * Holy Wu has added sized deallocation unit tests.
- * Holy Wu has enabled support of sized deallocation (c++14) on recent MSVC.
- * Holy Wu has fixed MSVC build in WIN32_OVERRIDE_ALLOCATORS mode. This closed
-   issue #716.
- * Holy Wu has contributed cleanup of config.h used on windows.
- * Mao Huang has contributed couple simple tcmalloc changes from chromium code
-   base. Making our tcmalloc forks a tiny bit closer.
- * issue #946 that caused compilation failures on some Linux clang installations
-   has been fixed. Much thanks to github user htuch for helping to diagnose
-   issue and proposing a fix.
- * Tulio Magno Quites Machado Filho has contributed build-time fix for PPC (for
-   problem introduced in one of commits since RC).</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8-1" date="1594036800">- gperftools 2.8 is out!
-
- Here are notable changes:
-
- * ProfilerGetStackTrace is now officially supported API for libprofiler.
-   Contributed by Kirill Müller.
- * Build failures on mingw were fixed. This fixed issue #1108.
- * Build failure of page_heap_test on MSVC was fixed.
- * Ryan Macnak contributed fix for compiling linux syscall support on i386 and
-   recent GCCs. This fixed issue #1076.
- * test failures caused by new gcc 10 optimizations were fixed. Same change also
-   fixed tests on clang.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8.1-1" date="1608552000">- gperftools-2.8.1 is out!
-
- Here are notable changes:
-
- * previous release contained change to release memory without page heap lock,
-   but this change had at least one bug that caused to crashes and corruption
-   when running under aggressive decommit mode (this is not default). While we
-   check for other bugs, this feature was reverted. See github issue #1204 and
-   issue #1227.
- * stack traces depth captured by gperftools is now up to 254 levels deep.
-   Thanks to Kerrick Staley for this small but useful tweak.
- * Levon Ter-Grigoryan has contributed small fix for compiler warning.
- * Grant Henke has contributed updated detection of program counter register for
-   OS X on arm64.
- * Tim Gates has contributed small typo fix.
- * Steve Langasek has contributed basic build fixes for riscv64.
- * Isaac Hier and okhowang have contributed premiliminary port of build
-   infrastructure to cmake. This works, but it is very premiliminary.
-   Autotools-based build is the only officially supported build for now.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9-1" date="1613908800">- gperftools 2.9 is out!
-
- Here are notable changes:
-
- * Jarno Rajahalme has contributed fix crashing bug in syscalls support for
-   aarch64.
- * User SSE4 has contributed basic support for Elbrus 2000 architecture (!)
- * Venkatesh Srinivas has contributed cleanup to atomic ops.
- * Đoàn Trần Công Danh has fixed cpu profiler compilation on musl.
- * there is now better backtracing support for aarch64 and riscv. x86-64 with
-   frame pointers now also defaults to this new "generic" frame pointer
-   backtracer.
- * emergency malloc is now enabled by default. Fixes hang on musl when libgcc
-   backtracer is enabled.
- * bunch of legacy config tests has been removed
- * Venkatesh Srinivas has contributed thread-safety annotations support.
- * couple more unit test bugs that caused tcmalloc_unittest to fail on recent
-   clang has been fixed.
- * usage of unsupportable linux_syscall_support.h has been removed from few
-   places. Building with --disable-heap-checker now completely avoids it. Expect
-   complete death of this header in next major release.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9.1-1" date="1614772800">- gperftools 2.9.1 is out!
-
- Minor fixes landed since previous release:
-
- * OSX builds now prefer backtrace() and have somewhat working heap sampling.
- * Incorrect assertion failure was fixed that crashed tcmalloc if assertions
-   were on and sized delete was used. More details in github issue #1254.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.10-1" date="1653912000">- gperftools 2.10 is out!
-
- Here are notable changes:
-
- * Matt T. Proud contributed documentation fix to call Go programming language
-   by it's true name instead of golang.
- * Robert Scott contributed debugallocator feature to use readable (PROT_READ)
-   fence pages. This is activated by TCMALLOC_PAGE_FENCE_READABLE environment
-   veriable.
- * User stdpain contributed fix for cmake detection of libunwind.
- * Natale Patriciello contributed fix for OSX Monterey support.
- * Volodymyr Nikolaichuk contributed support for returning memory back to OS by
-   using mmap with MAP_FIXED and PROT_NONE. It is off by default and enabled by
-   preprocessor define: FREE_MMAP_PROT_NONE. This should help OSes that don't
-   support Linux-style madvise MADV_DONTNEED or BSD-style MADV_FREE.
- * Jingyun Hua has contributed basic support for LoongArch.
- * Github issue #1338 of failing to build on some recent musl versions has been
-   fixed.
- * Github issue #1321 of failing to ship cmake bits with .tar.gz archive has
-   been fixed.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.11-1" date="1692014400">- gperftools 2.11 is out!
-
- Most notable change is that Linux/aarch64 and Linux/riscv are now fully
- supported. That is, all unit tests pass on those architectures (previously the
- heap leak checker was broken).
-
- Also notable is that heap leak checker support is officially deprecated as of
- this release. All bug fixes from now are on a best effort basis. For clarity we
- also declare that it is only expected to work (for some definition of work) on
- Linux/x86 (all kinds), Linux/aarch64, Linux/arm, Linux/ppc (untested as of this
- writing) and Linux/mips (untested as well). While some functionality worked in
- the past on BSDs, it was never fully functional; and will never be. We strongly
- recommend everyone to switch to asan and friends.
-
- For major internal changes it is also worth mentioning that we now fully
- switched to C++-11 std::atomic. All custom OS- and arch-specific atomic bits
- have been removed at last.
-
- Another notable change is that mmap and sbrk hooks facility is now no-op. We
- keep API and ABI for formal compatibility, but the calls to add mmap/sbrk hooks
- do nothing and return an error (whenever possible as part of API). There seem
- to be no users of it anyways, and mmap replacement API that is part of that
- facility really screwed up 64-bit offsets on (some/most) 32-bit systems.
- Internally for heap profiler and heap checker we have a new, but non-public API
- (see mmap_hook.h).
-
- Most tests now pass on NetBSD x86-64 (I tested on version 9.2). And only one
- that fails is new stacktrace test for stacktraces from signal handler (so there
- could be some imperfections for cpu profiles).
-
- We don't warn people away from the libgcc stacktrace capturing method anymore.
- In fact users on most recent glibc-s are advised to use it (pass
- --enable-libgcc-unwinder-by-default). This is thanks to the dl_find_object API
- offered by glibc which allows this implementation to be fully
- async-signal-safe. Modern Linux distros should from now on build their
- gperftools package with this enabled (other than those built on top of musl).
-
- generic_fp and generic_fp_unsafe stacktrace capturing methods have been
- expanded for more architectures and even some basic non-Linux support. We have
- completely removed old x86-specific frame pointer stacktrace implementation in
- favor of those 2. _unsafe one should be roughly equivalent to the old x86
- method. And 'safe' one is recommended as a new default for those who want
- FP-based stacktracing. Safe implementation robustly checks memory before
- accessing it, preventing unlikely, but not impossible crashes when frame
- pointers are bogus.
-
- On platforms that support it, we now build gperftools with
- "-fno-omit-frame-pointer -momit-leaf-frame-pointer". This makes gperftools
- mostly frame-pointer-ful, but without performance hit in places that matter
- (this is how Google builds their binaries BTW). That should cover gcc (at
- least) on x86, aarch64 and riscv. Intention for this change is to make
- distro-shipped libtcmalloc.so compatible with frame-pointer stacktrace
- capturing (for those who still do heap profiling, for example). Of course,
- passing --enable-frame-pointers still gives you full frame pointers (i.e. even
- for leaf functions).
-
- There is now support for detecting actual page size at runtime. tcmalloc will
- now allocate memory in units of this page size. It particularly helps on arms
- with 64k pages to return memory back to the kernel. But it is somewhat
- controversial, because it effectively bumps tcmalloc logical page size on those
- machines potentially increasing fragmentation. In any case, there is now a new
- environment variable TCMALLOC_OVERRIDE_PAGESIZE allowing people to override
- this check. I.e. to either reduce effective page size down to tcmalloc's
- logical page size or to increase it.
-
- MallocExtension::MarkThreadTemporarilyIdle has been changed to be identical to
- MarkThreadIdle. MarkThreadTemporarilyIdle is believed to be unused, anyways.
- See issue #880 for details.
-
- There are a whole bunch of smaller fixes. Many of those smaller fixes had no
- associated ticket, but some had. People are advised to see here for list of
- notable tickets closed in this release:
- https://github.com/gperftools/gperftools/issues?q=label%3Afixed-in-2.11+
-
- Some of those tickets are quite notable (fixes for rare deadlocks in cpu
- profiler ProfilerStop or while capturing heap growth stacktraces (aka
- growthz)).
-
- Here is list of notable contributions:
-
- * Chris Cambly has contributed initial support for AIX
- * Ali Saidi has contributed SpinlockPause implementation for aarch64
- * Henrik Reinstädtler has contributed fix for cpuprofiler on aarch64 OSX
- * Gabriel Marin has backported Chromium's commit for always sanity checking
-   large frees
- * User zhangyiru has contributed a fix to report the number of leaked bytes as
-   size_t instead of (usually 32-bit) int.
- * Sergey Fedorov has contributed some fix for building on older ppc-based
-   OSX-es
- * User tigeran has removed unused using declaration
- * Artem Polyakov has contributed auto-detection of several MPI systems w.r.t.
-   filenames used by HEAPPROFILE and CPUPROFILE environment variables. Also, we
-   now support HEAPPROFILE_USE_PID and CPUPROFILE_USE_PID environment variables
-   that force profile filenames to have pid appended. Which will be useful for
-   some programs that fork for parallelism. See #1263 for details.
- * Ken Raffenetti has extended MPI detection mentioned above with detection of
-   MPICH system.
-
- Thanks a lot!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.12-1" date="1692878400">- Brett T. Warden contributed one significant fix. After a change in the previous
- release, we installed broken pkg-config files. Brett noticed and fixed that.
- Huge thanks!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.13-1" date="1694433600">- gperftools 2.13 is out!
-
- This release includes a few minor fixes:
-
- * Ivan Dlugos has fixed some issues with cmake and config.h defines.
- * 32-bit builds no longer require 64-bit atomics (which we wrongly introduced
-   in 2.11 and which broke builds on some 32-bit architectures).
- * generic_fp backtracing method now uses robust address probing method. The
-   previous approach had occasional false positives, which caused occasional
-   rare crashes.
- * In some cases, MSVC generated TrivialOnce machine code that deadlocked
-   programs on startup. The issue is now fixed.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.14-1" date="1704110400">- gperftools 2.14 is out!
-
- This release has the following set of notable changes:
-
- * Roman Geissler has contributed a fix to nasty initialization bug introduced
-   in 2.13 (see github issue #1452 for one example where it fails).
- * spinlock delay support now has proper windows support. Instead of simply
-   sleeping, it uses WaitOnAddress (which is basically windows equivalent of
-   futexes). This improvement was contributed by Lennox Ho.
- * we now have basic QNX support (basic malloc + heap profiler) championed by
-   Xiang.Lin. Thanks! Do note, however, that QNX doesn't provide SIGPROF ticks,
-   so there will be no cpu profiler support on this OS.
- * Yikai Zhao has contributed several fixes to important corner cases of
-   generic_fp stacktrace method.
- * several people have contributed various improvements to our cmake build:
-   Lennox Ho, Sergey Fedorov, Mateusz Jakub Fila. But do note that cmake build
-   is still incomplete and best-effort.
- * Julian Schroeder have fixed generic_fp incompatibility with ARM pointer auth.
- * Mateusz Jakub Fila has contributed implementation of mallocinfo2 function
-   (64-bit version of mallinfo).
- * Lennox Ho has updated C malloc extension shims to include
-   {Set,Get}MemoryReleaseRate.
- * Lennox Ho has contributed the ability to disable malloc functions patching on
-   windows when TCMALLOC_DISABLE_REPLACEMENT=1 environment variable is set.
- * User poljak181 has contributed a fix to infinite recursion in some cases of
-   malloc hooks (or user-replaced operator new) and MallocExtension::instance().
- * Sergey Fedorov has contributed a fix to use MAP_ANON on some older OSes
-   without MAP_ANONYMOUS.
- * the way we detect working ucontext-&gt;pc extraction method was reworked and is
-   now fully compile-time as opposed to config-time. This means no more
-   duplication and mismatches between autoconf and cmake bits in this area.
-
- List of relevant tickets can be seen online at:
- https://github.com/gperftools/gperftools/issues?q=label%3Afixed-in-2.14+</changelog>
</package>
<package pkgid="31c2c786d7016ec99b5ef39be9b391f3bfcfca3aa1e556c36f7cb6a3ce317087" name="gperftools" arch="src">
  <version epoch="0" ver="2.15" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8-1" date="1594036800">- gperftools 2.8 is out!
-
- Here are notable changes:
-
- * ProfilerGetStackTrace is now officially supported API for libprofiler.
-   Contributed by Kirill Müller.
- * Build failures on mingw were fixed. This fixed issue #1108.
- * Build failure of page_heap_test on MSVC was fixed.
- * Ryan Macnak contributed fix for compiling linux syscall support on i386 and
-   recent GCCs. This fixed issue #1076.
- * test failures caused by new gcc 10 optimizations were fixed. Same change also
-   fixed tests on clang.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8.1-1" date="1608552000">- gperftools-2.8.1 is out!
-
- Here are notable changes:
-
- * previous release contained change to release memory without page heap lock,
-   but this change had at least one bug that caused to crashes and corruption
-   when running under aggressive decommit mode (this is not default). While we
-   check for other bugs, this feature was reverted. See github issue #1204 and
-   issue #1227.
- * stack traces depth captured by gperftools is now up to 254 levels deep.
-   Thanks to Kerrick Staley for this small but useful tweak.
- * Levon Ter-Grigoryan has contributed small fix for compiler warning.
- * Grant Henke has contributed updated detection of program counter register for
-   OS X on arm64.
- * Tim Gates has contributed small typo fix.
- * Steve Langasek has contributed basic build fixes for riscv64.
- * Isaac Hier and okhowang have contributed premiliminary port of build
-   infrastructure to cmake. This works, but it is very premiliminary.
-   Autotools-based build is the only officially supported build for now.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9-1" date="1613908800">- gperftools 2.9 is out!
-
- Here are notable changes:
-
- * Jarno Rajahalme has contributed fix crashing bug in syscalls support for
-   aarch64.
- * User SSE4 has contributed basic support for Elbrus 2000 architecture (!)
- * Venkatesh Srinivas has contributed cleanup to atomic ops.
- * Đoàn Trần Công Danh has fixed cpu profiler compilation on musl.
- * there is now better backtracing support for aarch64 and riscv. x86-64 with
-   frame pointers now also defaults to this new "generic" frame pointer
-   backtracer.
- * emergency malloc is now enabled by default. Fixes hang on musl when libgcc
-   backtracer is enabled.
- * bunch of legacy config tests has been removed
- * Venkatesh Srinivas has contributed thread-safety annotations support.
- * couple more unit test bugs that caused tcmalloc_unittest to fail on recent
-   clang has been fixed.
- * usage of unsupportable linux_syscall_support.h has been removed from few
-   places. Building with --disable-heap-checker now completely avoids it. Expect
-   complete death of this header in next major release.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9.1-1" date="1614772800">- gperftools 2.9.1 is out!
-
- Minor fixes landed since previous release:
-
- * OSX builds now prefer backtrace() and have somewhat working heap sampling.
- * Incorrect assertion failure was fixed that crashed tcmalloc if assertions
-   were on and sized delete was used. More details in github issue #1254.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.10-1" date="1653912000">- gperftools 2.10 is out!
-
- Here are notable changes:
-
- * Matt T. Proud contributed documentation fix to call Go programming language
-   by it's true name instead of golang.
- * Robert Scott contributed debugallocator feature to use readable (PROT_READ)
-   fence pages. This is activated by TCMALLOC_PAGE_FENCE_READABLE environment
-   veriable.
- * User stdpain contributed fix for cmake detection of libunwind.
- * Natale Patriciello contributed fix for OSX Monterey support.
- * Volodymyr Nikolaichuk contributed support for returning memory back to OS by
-   using mmap with MAP_FIXED and PROT_NONE. It is off by default and enabled by
-   preprocessor define: FREE_MMAP_PROT_NONE. This should help OSes that don't
-   support Linux-style madvise MADV_DONTNEED or BSD-style MADV_FREE.
- * Jingyun Hua has contributed basic support for LoongArch.
- * Github issue #1338 of failing to build on some recent musl versions has been
-   fixed.
- * Github issue #1321 of failing to ship cmake bits with .tar.gz archive has
-   been fixed.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.11-1" date="1692014400">- gperftools 2.11 is out!
-
- Most notable change is that Linux/aarch64 and Linux/riscv are now fully
- supported. That is, all unit tests pass on those architectures (previously the
- heap leak checker was broken).
-
- Also notable is that heap leak checker support is officially deprecated as of
- this release. All bug fixes from now are on a best effort basis. For clarity we
- also declare that it is only expected to work (for some definition of work) on
- Linux/x86 (all kinds), Linux/aarch64, Linux/arm, Linux/ppc (untested as of this
- writing) and Linux/mips (untested as well). While some functionality worked in
- the past on BSDs, it was never fully functional; and will never be. We strongly
- recommend everyone to switch to asan and friends.
-
- For major internal changes it is also worth mentioning that we now fully
- switched to C++-11 std::atomic. All custom OS- and arch-specific atomic bits
- have been removed at last.
-
- Another notable change is that mmap and sbrk hooks facility is now no-op. We
- keep API and ABI for formal compatibility, but the calls to add mmap/sbrk hooks
- do nothing and return an error (whenever possible as part of API). There seem
- to be no users of it anyways, and mmap replacement API that is part of that
- facility really screwed up 64-bit offsets on (some/most) 32-bit systems.
- Internally for heap profiler and heap checker we have a new, but non-public API
- (see mmap_hook.h).
-
- Most tests now pass on NetBSD x86-64 (I tested on version 9.2). And only one
- that fails is new stacktrace test for stacktraces from signal handler (so there
- could be some imperfections for cpu profiles).
-
- We don't warn people away from the libgcc stacktrace capturing method anymore.
- In fact users on most recent glibc-s are advised to use it (pass
- --enable-libgcc-unwinder-by-default). This is thanks to the dl_find_object API
- offered by glibc which allows this implementation to be fully
- async-signal-safe. Modern Linux distros should from now on build their
- gperftools package with this enabled (other than those built on top of musl).
-
- generic_fp and generic_fp_unsafe stacktrace capturing methods have been
- expanded for more architectures and even some basic non-Linux support. We have
- completely removed old x86-specific frame pointer stacktrace implementation in
- favor of those 2. _unsafe one should be roughly equivalent to the old x86
- method. And 'safe' one is recommended as a new default for those who want
- FP-based stacktracing. Safe implementation robustly checks memory before
- accessing it, preventing unlikely, but not impossible crashes when frame
- pointers are bogus.
-
- On platforms that support it, we now build gperftools with
- "-fno-omit-frame-pointer -momit-leaf-frame-pointer". This makes gperftools
- mostly frame-pointer-ful, but without performance hit in places that matter
- (this is how Google builds their binaries BTW). That should cover gcc (at
- least) on x86, aarch64 and riscv. Intention for this change is to make
- distro-shipped libtcmalloc.so compatible with frame-pointer stacktrace
- capturing (for those who still do heap profiling, for example). Of course,
- passing --enable-frame-pointers still gives you full frame pointers (i.e. even
- for leaf functions).
-
- There is now support for detecting actual page size at runtime. tcmalloc will
- now allocate memory in units of this page size. It particularly helps on arms
- with 64k pages to return memory back to the kernel. But it is somewhat
- controversial, because it effectively bumps tcmalloc logical page size on those
- machines potentially increasing fragmentation. In any case, there is now a new
- environment variable TCMALLOC_OVERRIDE_PAGESIZE allowing people to override
- this check. I.e. to either reduce effective page size down to tcmalloc's
- logical page size or to increase it.
-
- MallocExtension::MarkThreadTemporarilyIdle has been changed to be identical to
- MarkThreadIdle. MarkThreadTemporarilyIdle is believed to be unused, anyways.
- See issue #880 for details.
-
- There are a whole bunch of smaller fixes. Many of those smaller fixes had no
- associated ticket, but some had. People are advised to see here for list of
- notable tickets closed in this release:
- https://github.com/gperftools/gperftools/issues?q=label%3Afixed-in-2.11+
-
- Some of those tickets are quite notable (fixes for rare deadlocks in cpu
- profiler ProfilerStop or while capturing heap growth stacktraces (aka
- growthz)).
-
- Here is list of notable contributions:
-
- * Chris Cambly has contributed initial support for AIX
- * Ali Saidi has contributed SpinlockPause implementation for aarch64
- * Henrik Reinstädtler has contributed fix for cpuprofiler on aarch64 OSX
- * Gabriel Marin has backported Chromium's commit for always sanity checking
-   large frees
- * User zhangyiru has contributed a fix to report the number of leaked bytes as
-   size_t instead of (usually 32-bit) int.
- * Sergey Fedorov has contributed some fix for building on older ppc-based
-   OSX-es
- * User tigeran has removed unused using declaration
- * Artem Polyakov has contributed auto-detection of several MPI systems w.r.t.
-   filenames used by HEAPPROFILE and CPUPROFILE environment variables. Also, we
-   now support HEAPPROFILE_USE_PID and CPUPROFILE_USE_PID environment variables
-   that force profile filenames to have pid appended. Which will be useful for
-   some programs that fork for parallelism. See #1263 for details.
- * Ken Raffenetti has extended MPI detection mentioned above with detection of
-   MPICH system.
-
- Thanks a lot!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.12-1" date="1692878400">- Brett T. Warden contributed one significant fix. After a change in the previous
- release, we installed broken pkg-config files. Brett noticed and fixed that.
- Huge thanks!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.13-1" date="1694433600">- gperftools 2.13 is out!
-
- This release includes a few minor fixes:
-
- * Ivan Dlugos has fixed some issues with cmake and config.h defines.
- * 32-bit builds no longer require 64-bit atomics (which we wrongly introduced
-   in 2.11 and which broke builds on some 32-bit architectures).
- * generic_fp backtracing method now uses robust address probing method. The
-   previous approach had occasional false positives, which caused occasional
-   rare crashes.
- * In some cases, MSVC generated TrivialOnce machine code that deadlocked
-   programs on startup. The issue is now fixed.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.14-1" date="1704110400">- gperftools 2.14 is out!
-
- This release has the following set of notable changes:
-
- * Roman Geissler has contributed a fix to nasty initialization bug introduced
-   in 2.13 (see github issue #1452 for one example where it fails).
- * spinlock delay support now has proper windows support. Instead of simply
-   sleeping, it uses WaitOnAddress (which is basically windows equivalent of
-   futexes). This improvement was contributed by Lennox Ho.
- * we now have basic QNX support (basic malloc + heap profiler) championed by
-   Xiang.Lin. Thanks! Do note, however, that QNX doesn't provide SIGPROF ticks,
-   so there will be no cpu profiler support on this OS.
- * Yikai Zhao has contributed several fixes to important corner cases of
-   generic_fp stacktrace method.
- * several people have contributed various improvements to our cmake build:
-   Lennox Ho, Sergey Fedorov, Mateusz Jakub Fila. But do note that cmake build
-   is still incomplete and best-effort.
- * Julian Schroeder have fixed generic_fp incompatibility with ARM pointer auth.
- * Mateusz Jakub Fila has contributed implementation of mallocinfo2 function
-   (64-bit version of mallinfo).
- * Lennox Ho has updated C malloc extension shims to include
-   {Set,Get}MemoryReleaseRate.
- * Lennox Ho has contributed the ability to disable malloc functions patching on
-   windows when TCMALLOC_DISABLE_REPLACEMENT=1 environment variable is set.
- * User poljak181 has contributed a fix to infinite recursion in some cases of
-   malloc hooks (or user-replaced operator new) and MallocExtension::instance().
- * Sergey Fedorov has contributed a fix to use MAP_ANON on some older OSes
-   without MAP_ANONYMOUS.
- * the way we detect working ucontext-&gt;pc extraction method was reworked and is
-   now fully compile-time as opposed to config-time. This means no more
-   duplication and mismatches between autoconf and cmake bits in this area.
-
- List of relevant tickets can be seen online at:
- https://github.com/gperftools/gperftools/issues?q=label%3Afixed-in-2.14+</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.15-1" date="1704456000">- This release has the following bug fixes:
-
- * Xiaowei Wang has pointed out the pthread linking issue on cmake on older
-   glibcs (where -pthread is not implicit). See #1473 for more details.
- * Mikael Simberg and Tom "spot" Callaway have pointed out the missing symbols
-   issue when linking PPC or i386 builds. #1474 has all the details.
-
- Huge thanks to all contributors!</changelog>
</package>
<package pkgid="7a36161ac110ecad29047c9659e236f437068e2ad81ae66984a97151801a757b" name="gperftools" arch="src">
  <version epoch="0" ver="2.16" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8.1-1" date="1608552000">- gperftools-2.8.1 is out!
-
- Here are notable changes:
-
- * previous release contained change to release memory without page heap lock,
-   but this change had at least one bug that caused to crashes and corruption
-   when running under aggressive decommit mode (this is not default). While we
-   check for other bugs, this feature was reverted. See github issue #1204 and
-   issue #1227.
- * stack traces depth captured by gperftools is now up to 254 levels deep.
-   Thanks to Kerrick Staley for this small but useful tweak.
- * Levon Ter-Grigoryan has contributed small fix for compiler warning.
- * Grant Henke has contributed updated detection of program counter register for
-   OS X on arm64.
- * Tim Gates has contributed small typo fix.
- * Steve Langasek has contributed basic build fixes for riscv64.
- * Isaac Hier and okhowang have contributed premiliminary port of build
-   infrastructure to cmake. This works, but it is very premiliminary.
-   Autotools-based build is the only officially supported build for now.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9-1" date="1613908800">- gperftools 2.9 is out!
-
- Here are notable changes:
-
- * Jarno Rajahalme has contributed fix crashing bug in syscalls support for
-   aarch64.
- * User SSE4 has contributed basic support for Elbrus 2000 architecture (!)
- * Venkatesh Srinivas has contributed cleanup to atomic ops.
- * Đoàn Trần Công Danh has fixed cpu profiler compilation on musl.
- * there is now better backtracing support for aarch64 and riscv. x86-64 with
-   frame pointers now also defaults to this new "generic" frame pointer
-   backtracer.
- * emergency malloc is now enabled by default. Fixes hang on musl when libgcc
-   backtracer is enabled.
- * bunch of legacy config tests has been removed
- * Venkatesh Srinivas has contributed thread-safety annotations support.
- * couple more unit test bugs that caused tcmalloc_unittest to fail on recent
-   clang has been fixed.
- * usage of unsupportable linux_syscall_support.h has been removed from few
-   places. Building with --disable-heap-checker now completely avoids it. Expect
-   complete death of this header in next major release.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9.1-1" date="1614772800">- gperftools 2.9.1 is out!
-
- Minor fixes landed since previous release:
-
- * OSX builds now prefer backtrace() and have somewhat working heap sampling.
- * Incorrect assertion failure was fixed that crashed tcmalloc if assertions
-   were on and sized delete was used. More details in github issue #1254.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.10-1" date="1653912000">- gperftools 2.10 is out!
-
- Here are notable changes:
-
- * Matt T. Proud contributed documentation fix to call Go programming language
-   by it's true name instead of golang.
- * Robert Scott contributed debugallocator feature to use readable (PROT_READ)
-   fence pages. This is activated by TCMALLOC_PAGE_FENCE_READABLE environment
-   veriable.
- * User stdpain contributed fix for cmake detection of libunwind.
- * Natale Patriciello contributed fix for OSX Monterey support.
- * Volodymyr Nikolaichuk contributed support for returning memory back to OS by
-   using mmap with MAP_FIXED and PROT_NONE. It is off by default and enabled by
-   preprocessor define: FREE_MMAP_PROT_NONE. This should help OSes that don't
-   support Linux-style madvise MADV_DONTNEED or BSD-style MADV_FREE.
- * Jingyun Hua has contributed basic support for LoongArch.
- * Github issue #1338 of failing to build on some recent musl versions has been
-   fixed.
- * Github issue #1321 of failing to ship cmake bits with .tar.gz archive has
-   been fixed.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.11-1" date="1692014400">- gperftools 2.11 is out!
-
- Most notable change is that Linux/aarch64 and Linux/riscv are now fully
- supported. That is, all unit tests pass on those architectures (previously the
- heap leak checker was broken).
-
- Also notable is that heap leak checker support is officially deprecated as of
- this release. All bug fixes from now are on a best effort basis. For clarity we
- also declare that it is only expected to work (for some definition of work) on
- Linux/x86 (all kinds), Linux/aarch64, Linux/arm, Linux/ppc (untested as of this
- writing) and Linux/mips (untested as well). While some functionality worked in
- the past on BSDs, it was never fully functional; and will never be. We strongly
- recommend everyone to switch to asan and friends.
-
- For major internal changes it is also worth mentioning that we now fully
- switched to C++-11 std::atomic. All custom OS- and arch-specific atomic bits
- have been removed at last.
-
- Another notable change is that mmap and sbrk hooks facility is now no-op. We
- keep API and ABI for formal compatibility, but the calls to add mmap/sbrk hooks
- do nothing and return an error (whenever possible as part of API). There seem
- to be no users of it anyways, and mmap replacement API that is part of that
- facility really screwed up 64-bit offsets on (some/most) 32-bit systems.
- Internally for heap profiler and heap checker we have a new, but non-public API
- (see mmap_hook.h).
-
- Most tests now pass on NetBSD x86-64 (I tested on version 9.2). And only one
- that fails is new stacktrace test for stacktraces from signal handler (so there
- could be some imperfections for cpu profiles).
-
- We don't warn people away from the libgcc stacktrace capturing method anymore.
- In fact users on most recent glibc-s are advised to use it (pass
- --enable-libgcc-unwinder-by-default). This is thanks to the dl_find_object API
- offered by glibc which allows this implementation to be fully
- async-signal-safe. Modern Linux distros should from now on build their
- gperftools package with this enabled (other than those built on top of musl).
-
- generic_fp and generic_fp_unsafe stacktrace capturing methods have been
- expanded for more architectures and even some basic non-Linux support. We have
- completely removed old x86-specific frame pointer stacktrace implementation in
- favor of those 2. _unsafe one should be roughly equivalent to the old x86
- method. And 'safe' one is recommended as a new default for those who want
- FP-based stacktracing. Safe implementation robustly checks memory before
- accessing it, preventing unlikely, but not impossible crashes when frame
- pointers are bogus.
-
- On platforms that support it, we now build gperftools with
- "-fno-omit-frame-pointer -momit-leaf-frame-pointer". This makes gperftools
- mostly frame-pointer-ful, but without performance hit in places that matter
- (this is how Google builds their binaries BTW). That should cover gcc (at
- least) on x86, aarch64 and riscv. Intention for this change is to make
- distro-shipped libtcmalloc.so compatible with frame-pointer stacktrace
- capturing (for those who still do heap profiling, for example). Of course,
- passing --enable-frame-pointers still gives you full frame pointers (i.e. even
- for leaf functions).
-
- There is now support for detecting actual page size at runtime. tcmalloc will
- now allocate memory in units of this page size. It particularly helps on arms
- with 64k pages to return memory back to the kernel. But it is somewhat
- controversial, because it effectively bumps tcmalloc logical page size on those
- machines potentially increasing fragmentation. In any case, there is now a new
- environment variable TCMALLOC_OVERRIDE_PAGESIZE allowing people to override
- this check. I.e. to either reduce effective page size down to tcmalloc's
- logical page size or to increase it.
-
- MallocExtension::MarkThreadTemporarilyIdle has been changed to be identical to
- MarkThreadIdle. MarkThreadTemporarilyIdle is believed to be unused, anyways.
- See issue #880 for details.
-
- There are a whole bunch of smaller fixes. Many of those smaller fixes had no
- associated ticket, but some had. People are advised to see here for list of
- notable tickets closed in this release:
- https://github.com/gperftools/gperftools/issues?q=label%3Afixed-in-2.11+
-
- Some of those tickets are quite notable (fixes for rare deadlocks in cpu
- profiler ProfilerStop or while capturing heap growth stacktraces (aka
- growthz)).
-
- Here is list of notable contributions:
-
- * Chris Cambly has contributed initial support for AIX
- * Ali Saidi has contributed SpinlockPause implementation for aarch64
- * Henrik Reinstädtler has contributed fix for cpuprofiler on aarch64 OSX
- * Gabriel Marin has backported Chromium's commit for always sanity checking
-   large frees
- * User zhangyiru has contributed a fix to report the number of leaked bytes as
-   size_t instead of (usually 32-bit) int.
- * Sergey Fedorov has contributed some fix for building on older ppc-based
-   OSX-es
- * User tigeran has removed unused using declaration
- * Artem Polyakov has contributed auto-detection of several MPI systems w.r.t.
-   filenames used by HEAPPROFILE and CPUPROFILE environment variables. Also, we
-   now support HEAPPROFILE_USE_PID and CPUPROFILE_USE_PID environment variables
-   that force profile filenames to have pid appended. Which will be useful for
-   some programs that fork for parallelism. See #1263 for details.
- * Ken Raffenetti has extended MPI detection mentioned above with detection of
-   MPICH system.
-
- Thanks a lot!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.12-1" date="1692878400">- Brett T. Warden contributed one significant fix. After a change in the previous
- release, we installed broken pkg-config files. Brett noticed and fixed that.
- Huge thanks!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.13-1" date="1694433600">- gperftools 2.13 is out!
-
- This release includes a few minor fixes:
-
- * Ivan Dlugos has fixed some issues with cmake and config.h defines.
- * 32-bit builds no longer require 64-bit atomics (which we wrongly introduced
-   in 2.11 and which broke builds on some 32-bit architectures).
- * generic_fp backtracing method now uses robust address probing method. The
-   previous approach had occasional false positives, which caused occasional
-   rare crashes.
- * In some cases, MSVC generated TrivialOnce machine code that deadlocked
-   programs on startup. The issue is now fixed.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.14-1" date="1704110400">- gperftools 2.14 is out!
-
- This release has the following set of notable changes:
-
- * Roman Geissler has contributed a fix to nasty initialization bug introduced
-   in 2.13 (see github issue #1452 for one example where it fails).
- * spinlock delay support now has proper windows support. Instead of simply
-   sleeping, it uses WaitOnAddress (which is basically windows equivalent of
-   futexes). This improvement was contributed by Lennox Ho.
- * we now have basic QNX support (basic malloc + heap profiler) championed by
-   Xiang.Lin. Thanks! Do note, however, that QNX doesn't provide SIGPROF ticks,
-   so there will be no cpu profiler support on this OS.
- * Yikai Zhao has contributed several fixes to important corner cases of
-   generic_fp stacktrace method.
- * several people have contributed various improvements to our cmake build:
-   Lennox Ho, Sergey Fedorov, Mateusz Jakub Fila. But do note that cmake build
-   is still incomplete and best-effort.
- * Julian Schroeder have fixed generic_fp incompatibility with ARM pointer auth.
- * Mateusz Jakub Fila has contributed implementation of mallocinfo2 function
-   (64-bit version of mallinfo).
- * Lennox Ho has updated C malloc extension shims to include
-   {Set,Get}MemoryReleaseRate.
- * Lennox Ho has contributed the ability to disable malloc functions patching on
-   windows when TCMALLOC_DISABLE_REPLACEMENT=1 environment variable is set.
- * User poljak181 has contributed a fix to infinite recursion in some cases of
-   malloc hooks (or user-replaced operator new) and MallocExtension::instance().
- * Sergey Fedorov has contributed a fix to use MAP_ANON on some older OSes
-   without MAP_ANONYMOUS.
- * the way we detect working ucontext-&gt;pc extraction method was reworked and is
-   now fully compile-time as opposed to config-time. This means no more
-   duplication and mismatches between autoconf and cmake bits in this area.
-
- List of relevant tickets can be seen online at:
- https://github.com/gperftools/gperftools/issues?q=label%3Afixed-in-2.14+</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.15-1" date="1704456000">- This release has the following bug fixes:
-
- * Xiaowei Wang has pointed out the pthread linking issue on cmake on older
-   glibcs (where -pthread is not implicit). See #1473 for more details.
- * Mikael Simberg and Tom "spot" Callaway have pointed out the missing symbols
-   issue when linking PPC or i386 builds. #1474 has all the details.
-
- Huge thanks to all contributors!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.16-1" date="1727265600">- gperftools 2.16 is out!
-
- This release doesn't have major fixes or big headline features, but it has
- quite a lot of internal modernizations and cleanups. By the number of commits,
- 2.16 is going to be our biggest release ever.
-
- This release's main focus was making our code and building infrastructure
- simpler, more straightforward, more portable, and more modern.
-
- Please note that the gperftools 2.16 release will be the last release with the
- heap leak checker included. The time has come to drop this feature entirely.
- All users should migrate to relevant gcc/clang sanitizers.
-
- Here are the most notable changes:
-
- * we've upgraded our C++ standard to C++ 17. Some fraction of our code base was
-   modernized.
- * We've integrated (vendored copy of) GoogleTest, and most tests now use it.
-   GoogleTest has helped us eliminate some legacy code and reduce the number of
-   tests that use shell scripts.
- * There are no more unnecessary wrappers around mutexes and threads for unit
-   tests. We now use C++ standard mutexes and threads in our tests.
- * We've done the bulk of the work necessary to enable hidden visibility. The
-   most significant change is that tests no longer reach into libtcmalloc's
-   guts. We use a special TestingPortal interface instead. We now offer the
-   --enable-hidden-visibility configure option, which does what it says. But
-   please note that hidden visibility is off by default for now.
- * autotools build was significantly refactored, modernized and simplified.
- * The cmake build has also been radically simplified. The previous version
-   attempted to duplicate the same complexity that we had in the autotools build
-   and did not do it very well. More tests now pass under cmake. But please note
-   that cmake support is still not entirely functional, and we're not yet able
-   to promise anything about it.
- * Thread-local storage access and emergency malloc integration have been
-   reworked. We now support emergency malloc even on systems with emutls and
-   similarly "bad" TLS support. As a result, backtracing is now more reliable
-   (e.g., on QNX).
- * OSX operator new/delete performance has been improved. OSX's malloc
-   performance is badly compromised by its support of malloc zones, so we cannot
-   help much (the same applies to much of our competition among memory
-   allocators). But the C++ new/delete API doesn't have to integrate with this
-   stuff, so we now directly replace those functions for a sizeable speedup.
-   Note that OSX performance is still not on par with other "prime tier" OSes
-   due to its lack of efficient TLS support.
- * Long deprecated google/ headers have been deleted (use, e.g.,
-   "gperftools/tcmalloc.h" instead)
- * All clang builds now use -Wthread-safety and actually check thread-safety
-   declarations
- * Our code has stopped being incompatible with _TIME_BITS=64 on modern GNU
-   Linux systems (relevant only for 32-bit systems)
- * OpenSolaris build has been verified and fixed when needed
-
- Thanks to the following people for code contributions:
-
- * Github user oPiZiL (build fix for gcc 7.5)
- * Github user zhangdexin (qnx fixes)
- * Ishant Goyal (support for configuring minimal per-thread cache size)
- * Lennox Ho (several build fixes and several fixes around Windows support)
- * Olivier Langlois
- * Sergey Fedorovhas (another fix for building gperftools on old PPC OSX
-   computers)
- * Xiang.Lin (several OSX fixes)
- * Yikai Zhao (aarch64 generic_fp stack frame validation)
-
- You can find the list of all GitHub issues fixes in this release here:
- https://github.com/gperftools/gperftools/issues?q=label%3Afixed-in-2.16+is%3Aclosed</changelog>
</package>
<package pkgid="b4a067669454bbf65b1c06509bc7059f5c1f0a137b0d4eee0eef70a87daae6e0" name="gperftools" arch="src">
  <version epoch="0" ver="2.6.1" rel="1.el8"/>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-1" date="1433246400">- Import to RHEL</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-2" date="1440590400">- gperf-allow-customizing-trace-filename.patch [bz#1232702]
- Resolves: bz#1232702
  (gperftools: tcmalloc debug version uses hard-coded path /tmp/google.alloc)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-7" date="1440590401">- Rebuild to fix NVR [bz#1269032]
- Resolves: bz#1269032
  (gperftools NVR lower than EPEL version)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1466596800">- gp-Use-initial-exec-tls-for-libunwind-s-recursion-flag.patch [bz#1339710]
- Resolves: bz#1339710
  (initalization of 'recursive' tls variable in libunwind stack capturer occasionally triggers deadlock in ceph)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1507809600">- Rebase to 2.6.1 [bz#1431240]
- Removed libunwind usage [bz#1467203]
- Resolves: bz#1431240 
  (gperftools fails to build on s390x, lacks s390x support)
- Resolves: bz#1467203
  (Please, remove libunwind from the gperftools-libs (and 389-ds-base) requirements)</changelog>
</package>
<package pkgid="c1023512cb75198c785804d14d085d2b058471dca1296c92700e5ed4aabb2a4a" name="gperftools" arch="src">
  <version epoch="0" ver="2.6.2" rel="1.el8"/>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-1" date="1433246400">- Import to RHEL</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-2" date="1440590400">- gperf-allow-customizing-trace-filename.patch [bz#1232702]
- Resolves: bz#1232702
  (gperftools: tcmalloc debug version uses hard-coded path /tmp/google.alloc)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-7" date="1440590401">- Rebuild to fix NVR [bz#1269032]
- Resolves: bz#1269032
  (gperftools NVR lower than EPEL version)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1466596800">- gp-Use-initial-exec-tls-for-libunwind-s-recursion-flag.patch [bz#1339710]
- Resolves: bz#1339710
  (initalization of 'recursive' tls variable in libunwind stack capturer occasionally triggers deadlock in ceph)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1507809600">- Rebase to 2.6.1 [bz#1431240]
- Removed libunwind usage [bz#1467203]
- Resolves: bz#1431240 
  (gperftools fails to build on s390x, lacks s390x support)
- Resolves: bz#1467203
  (Please, remove libunwind from the gperftools-libs (and 389-ds-base) requirements)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.2-1" date="1512043200">- gperftools 2.6.2 is out!
-
- Most notable change is recently added support for C++17 over-aligned allocation
- operators contributed by Andrey Semashev. I've extended his implementation to
- have roughly same performance as malloc/new. This release also has native
- support for C11 aligned_alloc.
-
- Rest is mostly bug fixes:
-
- * Jianbo Yang has contributed a fix for potentially severe data raceintroduced
-   by malloc fast-path work in gperftools 2.6. This race could cause occasional
-   violation of total thread cache size constraint. See issue #929 for more
-   details.
- * Correct behavior in out-of-memory condition in fast-path cases was restored.
-   This was another bug introduced by fast-path optimization in gperftools 2.6
-   which caused operator new to silently return NULL instead of doing correct
-   C++ OOM handling (calling new_handler and throwing bad_alloc).
- * Khem Raj has contributed couple build fixes for newer glibcs (ucontext_t vs
-   struct ucontext and loff_t definition)
- * Piotr Sikora has contributed build fix for OSX (not building unwind
-   benchmark). This was issue #910 (thanks to Yuriy Solovyov for reporting it).
- * Dorin Lazăr has contributed fix for compiler warning
- * issue #912 (occasional deadlocking calling getenv too early on windows) was
-   fixed. Thanks to github user shangcangriluo for reporting it.
- * Couple earlier lsan-related commits still causing occasional issues linking
-   on OSX has been reverted. See issue #901.
- * Volodimir Krylov has contributed GetProgramInvocationName for FreeBSD
- * changsu lee has contributed couple minor correctness fixes (missing va_end()
-   and missing free() call in rarely executed Symbolize path)
- * Andrew C. Morrow has contributed some more page heap stats. See issue #935.
- * some cases of built-time warnings from various gcc/clang versions about
-   throw() declarations have been fixes.</changelog>
</package>
<package pkgid="03918b9d8873a4cbea93ad3aa02b828e219615d4558fd0a78d5cf4715ecdc4d1" name="gperftools" arch="src">
  <version epoch="0" ver="2.6.3" rel="1.el8"/>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-1" date="1433246400">- Import to RHEL</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-2" date="1440590400">- gperf-allow-customizing-trace-filename.patch [bz#1232702]
- Resolves: bz#1232702
  (gperftools: tcmalloc debug version uses hard-coded path /tmp/google.alloc)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-7" date="1440590401">- Rebuild to fix NVR [bz#1269032]
- Resolves: bz#1269032
  (gperftools NVR lower than EPEL version)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1466596800">- gp-Use-initial-exec-tls-for-libunwind-s-recursion-flag.patch [bz#1339710]
- Resolves: bz#1339710
  (initalization of 'recursive' tls variable in libunwind stack capturer occasionally triggers deadlock in ceph)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1507809600">- Rebase to 2.6.1 [bz#1431240]
- Removed libunwind usage [bz#1467203]
- Resolves: bz#1431240 
  (gperftools fails to build on s390x, lacks s390x support)
- Resolves: bz#1467203
  (Please, remove libunwind from the gperftools-libs (and 389-ds-base) requirements)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.2-1" date="1512043200">- gperftools 2.6.2 is out!
-
- Most notable change is recently added support for C++17 over-aligned allocation
- operators contributed by Andrey Semashev. I've extended his implementation to
- have roughly same performance as malloc/new. This release also has native
- support for C11 aligned_alloc.
-
- Rest is mostly bug fixes:
-
- * Jianbo Yang has contributed a fix for potentially severe data raceintroduced
-   by malloc fast-path work in gperftools 2.6. This race could cause occasional
-   violation of total thread cache size constraint. See issue #929 for more
-   details.
- * Correct behavior in out-of-memory condition in fast-path cases was restored.
-   This was another bug introduced by fast-path optimization in gperftools 2.6
-   which caused operator new to silently return NULL instead of doing correct
-   C++ OOM handling (calling new_handler and throwing bad_alloc).
- * Khem Raj has contributed couple build fixes for newer glibcs (ucontext_t vs
-   struct ucontext and loff_t definition)
- * Piotr Sikora has contributed build fix for OSX (not building unwind
-   benchmark). This was issue #910 (thanks to Yuriy Solovyov for reporting it).
- * Dorin Lazăr has contributed fix for compiler warning
- * issue #912 (occasional deadlocking calling getenv too early on windows) was
-   fixed. Thanks to github user shangcangriluo for reporting it.
- * Couple earlier lsan-related commits still causing occasional issues linking
-   on OSX has been reverted. See issue #901.
- * Volodimir Krylov has contributed GetProgramInvocationName for FreeBSD
- * changsu lee has contributed couple minor correctness fixes (missing va_end()
-   and missing free() call in rarely executed Symbolize path)
- * Andrew C. Morrow has contributed some more page heap stats. See issue #935.
- * some cases of built-time warnings from various gcc/clang versions about
-   throw() declarations have been fixes.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.3-1" date="1512820800">- gperftools 2.6.3 is out!
-
- Just two fixes were made in this release:
-
- * Stephan Zuercher has contributed a build fix for some recent XCode versions.
-   See issue #942 for more details.
- * assertion failure on some windows builds introduced by 2.6.2 was fixed.
-   Thanks to github user nkeemik for reporting it and testing fix. See issue
-   #944 for more details.</changelog>
</package>
<package pkgid="1f8a9cfec3cc1d1dd3a59d8908ea7f33c49169b617bb0de6a38a187c1231ff35" name="gperftools" arch="src">
  <version epoch="0" ver="2.7" rel="1.el8"/>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-1" date="1433246400">- Import to RHEL</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-2" date="1440590400">- gperf-allow-customizing-trace-filename.patch [bz#1232702]
- Resolves: bz#1232702
  (gperftools: tcmalloc debug version uses hard-coded path /tmp/google.alloc)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-7" date="1440590401">- Rebuild to fix NVR [bz#1269032]
- Resolves: bz#1269032
  (gperftools NVR lower than EPEL version)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1466596800">- gp-Use-initial-exec-tls-for-libunwind-s-recursion-flag.patch [bz#1339710]
- Resolves: bz#1339710
  (initalization of 'recursive' tls variable in libunwind stack capturer occasionally triggers deadlock in ceph)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1507809600">- Rebase to 2.6.1 [bz#1431240]
- Removed libunwind usage [bz#1467203]
- Resolves: bz#1431240 
  (gperftools fails to build on s390x, lacks s390x support)
- Resolves: bz#1467203
  (Please, remove libunwind from the gperftools-libs (and 389-ds-base) requirements)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.2-1" date="1512043200">- gperftools 2.6.2 is out!
-
- Most notable change is recently added support for C++17 over-aligned allocation
- operators contributed by Andrey Semashev. I've extended his implementation to
- have roughly same performance as malloc/new. This release also has native
- support for C11 aligned_alloc.
-
- Rest is mostly bug fixes:
-
- * Jianbo Yang has contributed a fix for potentially severe data raceintroduced
-   by malloc fast-path work in gperftools 2.6. This race could cause occasional
-   violation of total thread cache size constraint. See issue #929 for more
-   details.
- * Correct behavior in out-of-memory condition in fast-path cases was restored.
-   This was another bug introduced by fast-path optimization in gperftools 2.6
-   which caused operator new to silently return NULL instead of doing correct
-   C++ OOM handling (calling new_handler and throwing bad_alloc).
- * Khem Raj has contributed couple build fixes for newer glibcs (ucontext_t vs
-   struct ucontext and loff_t definition)
- * Piotr Sikora has contributed build fix for OSX (not building unwind
-   benchmark). This was issue #910 (thanks to Yuriy Solovyov for reporting it).
- * Dorin Lazăr has contributed fix for compiler warning
- * issue #912 (occasional deadlocking calling getenv too early on windows) was
-   fixed. Thanks to github user shangcangriluo for reporting it.
- * Couple earlier lsan-related commits still causing occasional issues linking
-   on OSX has been reverted. See issue #901.
- * Volodimir Krylov has contributed GetProgramInvocationName for FreeBSD
- * changsu lee has contributed couple minor correctness fixes (missing va_end()
-   and missing free() call in rarely executed Symbolize path)
- * Andrew C. Morrow has contributed some more page heap stats. See issue #935.
- * some cases of built-time warnings from various gcc/clang versions about
-   throw() declarations have been fixes.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.3-1" date="1512820800">- gperftools 2.6.3 is out!
-
- Just two fixes were made in this release:
-
- * Stephan Zuercher has contributed a build fix for some recent XCode versions.
-   See issue #942 for more details.
- * assertion failure on some windows builds introduced by 2.6.2 was fixed.
-   Thanks to github user nkeemik for reporting it and testing fix. See issue
-   #944 for more details.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.7-1" date="1525089600">- gperftools 2.7 is out!
-
- Few people contributed minor, but important fixes since rc.
-
- Changes:
-
- * bug in span stats printing introduced by new scalable page heap change was
-   fixed.
- * Christoph Müllner has contributed couple warnings fixes and initial support
-   for aarch64_ilp32 architecture.
- * Ben Dang contributed documentation fix for heap checker.
- * Fabrice Fontaine contributed fixed for linking benchmarks with
-   --disable-static.
- * Holy Wu has added sized deallocation unit tests.
- * Holy Wu has enabled support of sized deallocation (c++14) on recent MSVC.
- * Holy Wu has fixed MSVC build in WIN32_OVERRIDE_ALLOCATORS mode. This closed
-   issue #716.
- * Holy Wu has contributed cleanup of config.h used on windows.
- * Mao Huang has contributed couple simple tcmalloc changes from chromium code
-   base. Making our tcmalloc forks a tiny bit closer.
- * issue #946 that caused compilation failures on some Linux clang installations
-   has been fixed. Much thanks to github user htuch for helping to diagnose
-   issue and proposing a fix.
- * Tulio Magno Quites Machado Filho has contributed build-time fix for PPC (for
-   problem introduced in one of commits since RC).</changelog>
</package>
<package pkgid="b31c3dd82f7a17e0a94884e3417c20a95132d6629c6e05065054abacbcf2fae9" name="gperftools" arch="src">
  <version epoch="0" ver="2.8" rel="1.el8"/>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-1" date="1433246400">- Import to RHEL</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-2" date="1440590400">- gperf-allow-customizing-trace-filename.patch [bz#1232702]
- Resolves: bz#1232702
  (gperftools: tcmalloc debug version uses hard-coded path /tmp/google.alloc)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-7" date="1440590401">- Rebuild to fix NVR [bz#1269032]
- Resolves: bz#1269032
  (gperftools NVR lower than EPEL version)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1466596800">- gp-Use-initial-exec-tls-for-libunwind-s-recursion-flag.patch [bz#1339710]
- Resolves: bz#1339710
  (initalization of 'recursive' tls variable in libunwind stack capturer occasionally triggers deadlock in ceph)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1507809600">- Rebase to 2.6.1 [bz#1431240]
- Removed libunwind usage [bz#1467203]
- Resolves: bz#1431240 
  (gperftools fails to build on s390x, lacks s390x support)
- Resolves: bz#1467203
  (Please, remove libunwind from the gperftools-libs (and 389-ds-base) requirements)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.2-1" date="1512043200">- gperftools 2.6.2 is out!
-
- Most notable change is recently added support for C++17 over-aligned allocation
- operators contributed by Andrey Semashev. I've extended his implementation to
- have roughly same performance as malloc/new. This release also has native
- support for C11 aligned_alloc.
-
- Rest is mostly bug fixes:
-
- * Jianbo Yang has contributed a fix for potentially severe data raceintroduced
-   by malloc fast-path work in gperftools 2.6. This race could cause occasional
-   violation of total thread cache size constraint. See issue #929 for more
-   details.
- * Correct behavior in out-of-memory condition in fast-path cases was restored.
-   This was another bug introduced by fast-path optimization in gperftools 2.6
-   which caused operator new to silently return NULL instead of doing correct
-   C++ OOM handling (calling new_handler and throwing bad_alloc).
- * Khem Raj has contributed couple build fixes for newer glibcs (ucontext_t vs
-   struct ucontext and loff_t definition)
- * Piotr Sikora has contributed build fix for OSX (not building unwind
-   benchmark). This was issue #910 (thanks to Yuriy Solovyov for reporting it).
- * Dorin Lazăr has contributed fix for compiler warning
- * issue #912 (occasional deadlocking calling getenv too early on windows) was
-   fixed. Thanks to github user shangcangriluo for reporting it.
- * Couple earlier lsan-related commits still causing occasional issues linking
-   on OSX has been reverted. See issue #901.
- * Volodimir Krylov has contributed GetProgramInvocationName for FreeBSD
- * changsu lee has contributed couple minor correctness fixes (missing va_end()
-   and missing free() call in rarely executed Symbolize path)
- * Andrew C. Morrow has contributed some more page heap stats. See issue #935.
- * some cases of built-time warnings from various gcc/clang versions about
-   throw() declarations have been fixes.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.3-1" date="1512820800">- gperftools 2.6.3 is out!
-
- Just two fixes were made in this release:
-
- * Stephan Zuercher has contributed a build fix for some recent XCode versions.
-   See issue #942 for more details.
- * assertion failure on some windows builds introduced by 2.6.2 was fixed.
-   Thanks to github user nkeemik for reporting it and testing fix. See issue
-   #944 for more details.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.7-1" date="1525089600">- gperftools 2.7 is out!
-
- Few people contributed minor, but important fixes since rc.
-
- Changes:
-
- * bug in span stats printing introduced by new scalable page heap change was
-   fixed.
- * Christoph Müllner has contributed couple warnings fixes and initial support
-   for aarch64_ilp32 architecture.
- * Ben Dang contributed documentation fix for heap checker.
- * Fabrice Fontaine contributed fixed for linking benchmarks with
-   --disable-static.
- * Holy Wu has added sized deallocation unit tests.
- * Holy Wu has enabled support of sized deallocation (c++14) on recent MSVC.
- * Holy Wu has fixed MSVC build in WIN32_OVERRIDE_ALLOCATORS mode. This closed
-   issue #716.
- * Holy Wu has contributed cleanup of config.h used on windows.
- * Mao Huang has contributed couple simple tcmalloc changes from chromium code
-   base. Making our tcmalloc forks a tiny bit closer.
- * issue #946 that caused compilation failures on some Linux clang installations
-   has been fixed. Much thanks to github user htuch for helping to diagnose
-   issue and proposing a fix.
- * Tulio Magno Quites Machado Filho has contributed build-time fix for PPC (for
-   problem introduced in one of commits since RC).</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8-1" date="1594036800">- gperftools 2.8 is out!
-
- Here are notable changes:
-
- * ProfilerGetStackTrace is now officially supported API for libprofiler.
-   Contributed by Kirill Müller.
- * Build failures on mingw were fixed. This fixed issue #1108.
- * Build failure of page_heap_test on MSVC was fixed.
- * Ryan Macnak contributed fix for compiling linux syscall support on i386 and
-   recent GCCs. This fixed issue #1076.
- * test failures caused by new gcc 10 optimizations were fixed. Same change also
-   fixed tests on clang.</changelog>
</package>
<package pkgid="e628776a71a32f5e38e6dc3879107552f2d3b113d8d822a9677718cdaa554f3c" name="gperftools" arch="src">
  <version epoch="0" ver="2.8.1" rel="1.el8"/>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-1" date="1433246400">- Import to RHEL</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-2" date="1440590400">- gperf-allow-customizing-trace-filename.patch [bz#1232702]
- Resolves: bz#1232702
  (gperftools: tcmalloc debug version uses hard-coded path /tmp/google.alloc)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-7" date="1440590401">- Rebuild to fix NVR [bz#1269032]
- Resolves: bz#1269032
  (gperftools NVR lower than EPEL version)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1466596800">- gp-Use-initial-exec-tls-for-libunwind-s-recursion-flag.patch [bz#1339710]
- Resolves: bz#1339710
  (initalization of 'recursive' tls variable in libunwind stack capturer occasionally triggers deadlock in ceph)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1507809600">- Rebase to 2.6.1 [bz#1431240]
- Removed libunwind usage [bz#1467203]
- Resolves: bz#1431240 
  (gperftools fails to build on s390x, lacks s390x support)
- Resolves: bz#1467203
  (Please, remove libunwind from the gperftools-libs (and 389-ds-base) requirements)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.2-1" date="1512043200">- gperftools 2.6.2 is out!
-
- Most notable change is recently added support for C++17 over-aligned allocation
- operators contributed by Andrey Semashev. I've extended his implementation to
- have roughly same performance as malloc/new. This release also has native
- support for C11 aligned_alloc.
-
- Rest is mostly bug fixes:
-
- * Jianbo Yang has contributed a fix for potentially severe data raceintroduced
-   by malloc fast-path work in gperftools 2.6. This race could cause occasional
-   violation of total thread cache size constraint. See issue #929 for more
-   details.
- * Correct behavior in out-of-memory condition in fast-path cases was restored.
-   This was another bug introduced by fast-path optimization in gperftools 2.6
-   which caused operator new to silently return NULL instead of doing correct
-   C++ OOM handling (calling new_handler and throwing bad_alloc).
- * Khem Raj has contributed couple build fixes for newer glibcs (ucontext_t vs
-   struct ucontext and loff_t definition)
- * Piotr Sikora has contributed build fix for OSX (not building unwind
-   benchmark). This was issue #910 (thanks to Yuriy Solovyov for reporting it).
- * Dorin Lazăr has contributed fix for compiler warning
- * issue #912 (occasional deadlocking calling getenv too early on windows) was
-   fixed. Thanks to github user shangcangriluo for reporting it.
- * Couple earlier lsan-related commits still causing occasional issues linking
-   on OSX has been reverted. See issue #901.
- * Volodimir Krylov has contributed GetProgramInvocationName for FreeBSD
- * changsu lee has contributed couple minor correctness fixes (missing va_end()
-   and missing free() call in rarely executed Symbolize path)
- * Andrew C. Morrow has contributed some more page heap stats. See issue #935.
- * some cases of built-time warnings from various gcc/clang versions about
-   throw() declarations have been fixes.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.3-1" date="1512820800">- gperftools 2.6.3 is out!
-
- Just two fixes were made in this release:
-
- * Stephan Zuercher has contributed a build fix for some recent XCode versions.
-   See issue #942 for more details.
- * assertion failure on some windows builds introduced by 2.6.2 was fixed.
-   Thanks to github user nkeemik for reporting it and testing fix. See issue
-   #944 for more details.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.7-1" date="1525089600">- gperftools 2.7 is out!
-
- Few people contributed minor, but important fixes since rc.
-
- Changes:
-
- * bug in span stats printing introduced by new scalable page heap change was
-   fixed.
- * Christoph Müllner has contributed couple warnings fixes and initial support
-   for aarch64_ilp32 architecture.
- * Ben Dang contributed documentation fix for heap checker.
- * Fabrice Fontaine contributed fixed for linking benchmarks with
-   --disable-static.
- * Holy Wu has added sized deallocation unit tests.
- * Holy Wu has enabled support of sized deallocation (c++14) on recent MSVC.
- * Holy Wu has fixed MSVC build in WIN32_OVERRIDE_ALLOCATORS mode. This closed
-   issue #716.
- * Holy Wu has contributed cleanup of config.h used on windows.
- * Mao Huang has contributed couple simple tcmalloc changes from chromium code
-   base. Making our tcmalloc forks a tiny bit closer.
- * issue #946 that caused compilation failures on some Linux clang installations
-   has been fixed. Much thanks to github user htuch for helping to diagnose
-   issue and proposing a fix.
- * Tulio Magno Quites Machado Filho has contributed build-time fix for PPC (for
-   problem introduced in one of commits since RC).</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8-1" date="1594036800">- gperftools 2.8 is out!
-
- Here are notable changes:
-
- * ProfilerGetStackTrace is now officially supported API for libprofiler.
-   Contributed by Kirill Müller.
- * Build failures on mingw were fixed. This fixed issue #1108.
- * Build failure of page_heap_test on MSVC was fixed.
- * Ryan Macnak contributed fix for compiling linux syscall support on i386 and
-   recent GCCs. This fixed issue #1076.
- * test failures caused by new gcc 10 optimizations were fixed. Same change also
-   fixed tests on clang.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8.1-1" date="1608552000">- gperftools-2.8.1 is out!
-
- Here are notable changes:
-
- * previous release contained change to release memory without page heap lock,
-   but this change had at least one bug that caused to crashes and corruption
-   when running under aggressive decommit mode (this is not default). While we
-   check for other bugs, this feature was reverted. See github issue #1204 and
-   issue #1227.
- * stack traces depth captured by gperftools is now up to 254 levels deep.
-   Thanks to Kerrick Staley for this small but useful tweak.
- * Levon Ter-Grigoryan has contributed small fix for compiler warning.
- * Grant Henke has contributed updated detection of program counter register for
-   OS X on arm64.
- * Tim Gates has contributed small typo fix.
- * Steve Langasek has contributed basic build fixes for riscv64.
- * Isaac Hier and okhowang have contributed premiliminary port of build
-   infrastructure to cmake. This works, but it is very premiliminary.
-   Autotools-based build is the only officially supported build for now.</changelog>
</package>
<package pkgid="ebd193e778a6cb2105f1fb852db868dcbc752a236c7cff7fa6796e49a70783e5" name="gperftools" arch="src">
  <version epoch="0" ver="2.9" rel="1.el8"/>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-2" date="1440590400">- gperf-allow-customizing-trace-filename.patch [bz#1232702]
- Resolves: bz#1232702
  (gperftools: tcmalloc debug version uses hard-coded path /tmp/google.alloc)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-7" date="1440590401">- Rebuild to fix NVR [bz#1269032]
- Resolves: bz#1269032
  (gperftools NVR lower than EPEL version)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1466596800">- gp-Use-initial-exec-tls-for-libunwind-s-recursion-flag.patch [bz#1339710]
- Resolves: bz#1339710
  (initalization of 'recursive' tls variable in libunwind stack capturer occasionally triggers deadlock in ceph)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1507809600">- Rebase to 2.6.1 [bz#1431240]
- Removed libunwind usage [bz#1467203]
- Resolves: bz#1431240 
  (gperftools fails to build on s390x, lacks s390x support)
- Resolves: bz#1467203
  (Please, remove libunwind from the gperftools-libs (and 389-ds-base) requirements)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.2-1" date="1512043200">- gperftools 2.6.2 is out!
-
- Most notable change is recently added support for C++17 over-aligned allocation
- operators contributed by Andrey Semashev. I've extended his implementation to
- have roughly same performance as malloc/new. This release also has native
- support for C11 aligned_alloc.
-
- Rest is mostly bug fixes:
-
- * Jianbo Yang has contributed a fix for potentially severe data raceintroduced
-   by malloc fast-path work in gperftools 2.6. This race could cause occasional
-   violation of total thread cache size constraint. See issue #929 for more
-   details.
- * Correct behavior in out-of-memory condition in fast-path cases was restored.
-   This was another bug introduced by fast-path optimization in gperftools 2.6
-   which caused operator new to silently return NULL instead of doing correct
-   C++ OOM handling (calling new_handler and throwing bad_alloc).
- * Khem Raj has contributed couple build fixes for newer glibcs (ucontext_t vs
-   struct ucontext and loff_t definition)
- * Piotr Sikora has contributed build fix for OSX (not building unwind
-   benchmark). This was issue #910 (thanks to Yuriy Solovyov for reporting it).
- * Dorin Lazăr has contributed fix for compiler warning
- * issue #912 (occasional deadlocking calling getenv too early on windows) was
-   fixed. Thanks to github user shangcangriluo for reporting it.
- * Couple earlier lsan-related commits still causing occasional issues linking
-   on OSX has been reverted. See issue #901.
- * Volodimir Krylov has contributed GetProgramInvocationName for FreeBSD
- * changsu lee has contributed couple minor correctness fixes (missing va_end()
-   and missing free() call in rarely executed Symbolize path)
- * Andrew C. Morrow has contributed some more page heap stats. See issue #935.
- * some cases of built-time warnings from various gcc/clang versions about
-   throw() declarations have been fixes.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.3-1" date="1512820800">- gperftools 2.6.3 is out!
-
- Just two fixes were made in this release:
-
- * Stephan Zuercher has contributed a build fix for some recent XCode versions.
-   See issue #942 for more details.
- * assertion failure on some windows builds introduced by 2.6.2 was fixed.
-   Thanks to github user nkeemik for reporting it and testing fix. See issue
-   #944 for more details.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.7-1" date="1525089600">- gperftools 2.7 is out!
-
- Few people contributed minor, but important fixes since rc.
-
- Changes:
-
- * bug in span stats printing introduced by new scalable page heap change was
-   fixed.
- * Christoph Müllner has contributed couple warnings fixes and initial support
-   for aarch64_ilp32 architecture.
- * Ben Dang contributed documentation fix for heap checker.
- * Fabrice Fontaine contributed fixed for linking benchmarks with
-   --disable-static.
- * Holy Wu has added sized deallocation unit tests.
- * Holy Wu has enabled support of sized deallocation (c++14) on recent MSVC.
- * Holy Wu has fixed MSVC build in WIN32_OVERRIDE_ALLOCATORS mode. This closed
-   issue #716.
- * Holy Wu has contributed cleanup of config.h used on windows.
- * Mao Huang has contributed couple simple tcmalloc changes from chromium code
-   base. Making our tcmalloc forks a tiny bit closer.
- * issue #946 that caused compilation failures on some Linux clang installations
-   has been fixed. Much thanks to github user htuch for helping to diagnose
-   issue and proposing a fix.
- * Tulio Magno Quites Machado Filho has contributed build-time fix for PPC (for
-   problem introduced in one of commits since RC).</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8-1" date="1594036800">- gperftools 2.8 is out!
-
- Here are notable changes:
-
- * ProfilerGetStackTrace is now officially supported API for libprofiler.
-   Contributed by Kirill Müller.
- * Build failures on mingw were fixed. This fixed issue #1108.
- * Build failure of page_heap_test on MSVC was fixed.
- * Ryan Macnak contributed fix for compiling linux syscall support on i386 and
-   recent GCCs. This fixed issue #1076.
- * test failures caused by new gcc 10 optimizations were fixed. Same change also
-   fixed tests on clang.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8.1-1" date="1608552000">- gperftools-2.8.1 is out!
-
- Here are notable changes:
-
- * previous release contained change to release memory without page heap lock,
-   but this change had at least one bug that caused to crashes and corruption
-   when running under aggressive decommit mode (this is not default). While we
-   check for other bugs, this feature was reverted. See github issue #1204 and
-   issue #1227.
- * stack traces depth captured by gperftools is now up to 254 levels deep.
-   Thanks to Kerrick Staley for this small but useful tweak.
- * Levon Ter-Grigoryan has contributed small fix for compiler warning.
- * Grant Henke has contributed updated detection of program counter register for
-   OS X on arm64.
- * Tim Gates has contributed small typo fix.
- * Steve Langasek has contributed basic build fixes for riscv64.
- * Isaac Hier and okhowang have contributed premiliminary port of build
-   infrastructure to cmake. This works, but it is very premiliminary.
-   Autotools-based build is the only officially supported build for now.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9-1" date="1613908800">- gperftools 2.9 is out!
-
- Here are notable changes:
-
- * Jarno Rajahalme has contributed fix crashing bug in syscalls support for
-   aarch64.
- * User SSE4 has contributed basic support for Elbrus 2000 architecture (!)
- * Venkatesh Srinivas has contributed cleanup to atomic ops.
- * Đoàn Trần Công Danh has fixed cpu profiler compilation on musl.
- * there is now better backtracing support for aarch64 and riscv. x86-64 with
-   frame pointers now also defaults to this new "generic" frame pointer
-   backtracer.
- * emergency malloc is now enabled by default. Fixes hang on musl when libgcc
-   backtracer is enabled.
- * bunch of legacy config tests has been removed
- * Venkatesh Srinivas has contributed thread-safety annotations support.
- * couple more unit test bugs that caused tcmalloc_unittest to fail on recent
-   clang has been fixed.
- * usage of unsupportable linux_syscall_support.h has been removed from few
-   places. Building with --disable-heap-checker now completely avoids it. Expect
-   complete death of this header in next major release.</changelog>
</package>
<package pkgid="b6299584da89e1e22d1969118f8cc0d008fd6774b9de6d3e654e55f0464cfe7e" name="gperftools" arch="src">
  <version epoch="0" ver="2.9.1" rel="1.el8"/>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; 2.4-7" date="1440590400">- Rebuild to fix NVR [bz#1269032]
- Resolves: bz#1269032
  (gperftools NVR lower than EPEL version)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1466596800">- gp-Use-initial-exec-tls-for-libunwind-s-recursion-flag.patch [bz#1339710]
- Resolves: bz#1339710
  (initalization of 'recursive' tls variable in libunwind stack capturer occasionally triggers deadlock in ceph)</changelog>
  <changelog author="Miroslav Rezanina &lt;mrezanin@redhat.com&gt; - 2.4-8.el7" date="1507809600">- Rebase to 2.6.1 [bz#1431240]
- Removed libunwind usage [bz#1467203]
- Resolves: bz#1431240 
  (gperftools fails to build on s390x, lacks s390x support)
- Resolves: bz#1467203
  (Please, remove libunwind from the gperftools-libs (and 389-ds-base) requirements)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.2-1" date="1512043200">- gperftools 2.6.2 is out!
-
- Most notable change is recently added support for C++17 over-aligned allocation
- operators contributed by Andrey Semashev. I've extended his implementation to
- have roughly same performance as malloc/new. This release also has native
- support for C11 aligned_alloc.
-
- Rest is mostly bug fixes:
-
- * Jianbo Yang has contributed a fix for potentially severe data raceintroduced
-   by malloc fast-path work in gperftools 2.6. This race could cause occasional
-   violation of total thread cache size constraint. See issue #929 for more
-   details.
- * Correct behavior in out-of-memory condition in fast-path cases was restored.
-   This was another bug introduced by fast-path optimization in gperftools 2.6
-   which caused operator new to silently return NULL instead of doing correct
-   C++ OOM handling (calling new_handler and throwing bad_alloc).
- * Khem Raj has contributed couple build fixes for newer glibcs (ucontext_t vs
-   struct ucontext and loff_t definition)
- * Piotr Sikora has contributed build fix for OSX (not building unwind
-   benchmark). This was issue #910 (thanks to Yuriy Solovyov for reporting it).
- * Dorin Lazăr has contributed fix for compiler warning
- * issue #912 (occasional deadlocking calling getenv too early on windows) was
-   fixed. Thanks to github user shangcangriluo for reporting it.
- * Couple earlier lsan-related commits still causing occasional issues linking
-   on OSX has been reverted. See issue #901.
- * Volodimir Krylov has contributed GetProgramInvocationName for FreeBSD
- * changsu lee has contributed couple minor correctness fixes (missing va_end()
-   and missing free() call in rarely executed Symbolize path)
- * Andrew C. Morrow has contributed some more page heap stats. See issue #935.
- * some cases of built-time warnings from various gcc/clang versions about
-   throw() declarations have been fixes.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.6.3-1" date="1512820800">- gperftools 2.6.3 is out!
-
- Just two fixes were made in this release:
-
- * Stephan Zuercher has contributed a build fix for some recent XCode versions.
-   See issue #942 for more details.
- * assertion failure on some windows builds introduced by 2.6.2 was fixed.
-   Thanks to github user nkeemik for reporting it and testing fix. See issue
-   #944 for more details.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.7-1" date="1525089600">- gperftools 2.7 is out!
-
- Few people contributed minor, but important fixes since rc.
-
- Changes:
-
- * bug in span stats printing introduced by new scalable page heap change was
-   fixed.
- * Christoph Müllner has contributed couple warnings fixes and initial support
-   for aarch64_ilp32 architecture.
- * Ben Dang contributed documentation fix for heap checker.
- * Fabrice Fontaine contributed fixed for linking benchmarks with
-   --disable-static.
- * Holy Wu has added sized deallocation unit tests.
- * Holy Wu has enabled support of sized deallocation (c++14) on recent MSVC.
- * Holy Wu has fixed MSVC build in WIN32_OVERRIDE_ALLOCATORS mode. This closed
-   issue #716.
- * Holy Wu has contributed cleanup of config.h used on windows.
- * Mao Huang has contributed couple simple tcmalloc changes from chromium code
-   base. Making our tcmalloc forks a tiny bit closer.
- * issue #946 that caused compilation failures on some Linux clang installations
-   has been fixed. Much thanks to github user htuch for helping to diagnose
-   issue and proposing a fix.
- * Tulio Magno Quites Machado Filho has contributed build-time fix for PPC (for
-   problem introduced in one of commits since RC).</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8-1" date="1594036800">- gperftools 2.8 is out!
-
- Here are notable changes:
-
- * ProfilerGetStackTrace is now officially supported API for libprofiler.
-   Contributed by Kirill Müller.
- * Build failures on mingw were fixed. This fixed issue #1108.
- * Build failure of page_heap_test on MSVC was fixed.
- * Ryan Macnak contributed fix for compiling linux syscall support on i386 and
-   recent GCCs. This fixed issue #1076.
- * test failures caused by new gcc 10 optimizations were fixed. Same change also
-   fixed tests on clang.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8.1-1" date="1608552000">- gperftools-2.8.1 is out!
-
- Here are notable changes:
-
- * previous release contained change to release memory without page heap lock,
-   but this change had at least one bug that caused to crashes and corruption
-   when running under aggressive decommit mode (this is not default). While we
-   check for other bugs, this feature was reverted. See github issue #1204 and
-   issue #1227.
- * stack traces depth captured by gperftools is now up to 254 levels deep.
-   Thanks to Kerrick Staley for this small but useful tweak.
- * Levon Ter-Grigoryan has contributed small fix for compiler warning.
- * Grant Henke has contributed updated detection of program counter register for
-   OS X on arm64.
- * Tim Gates has contributed small typo fix.
- * Steve Langasek has contributed basic build fixes for riscv64.
- * Isaac Hier and okhowang have contributed premiliminary port of build
-   infrastructure to cmake. This works, but it is very premiliminary.
-   Autotools-based build is the only officially supported build for now.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9-1" date="1613908800">- gperftools 2.9 is out!
-
- Here are notable changes:
-
- * Jarno Rajahalme has contributed fix crashing bug in syscalls support for
-   aarch64.
- * User SSE4 has contributed basic support for Elbrus 2000 architecture (!)
- * Venkatesh Srinivas has contributed cleanup to atomic ops.
- * Đoàn Trần Công Danh has fixed cpu profiler compilation on musl.
- * there is now better backtracing support for aarch64 and riscv. x86-64 with
-   frame pointers now also defaults to this new "generic" frame pointer
-   backtracer.
- * emergency malloc is now enabled by default. Fixes hang on musl when libgcc
-   backtracer is enabled.
- * bunch of legacy config tests has been removed
- * Venkatesh Srinivas has contributed thread-safety annotations support.
- * couple more unit test bugs that caused tcmalloc_unittest to fail on recent
-   clang has been fixed.
- * usage of unsupportable linux_syscall_support.h has been removed from few
-   places. Building with --disable-heap-checker now completely avoids it. Expect
-   complete death of this header in next major release.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.9.1-1" date="1614772800">- gperftools 2.9.1 is out!
-
- Minor fixes landed since previous release:
-
- * OSX builds now prefer backtrace() and have somewhat working heap sampling.
- * Incorrect assertion failure was fixed that crashed tcmalloc if assertions
-   were on and sized delete was used. More details in github issue #1254.</changelog>
</package>
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  <version epoch="1" ver="1.3.4" rel="5.el8"/>
  <changelog author="Steve Traylen &lt;steve.traylen@cern.ch&gt; - 1.3.2-1" date="1400068800">- New version 1.3.2</changelog>
  <changelog author="Steve Traylen &lt;steve.traylen@cern.ch&gt; - 1.3.2-2" date="1400068801">- Packaging error</changelog>
  <changelog author="Steve Traylen &lt;steve.traylen@cern.ch&gt; - 1.3.3-1" date="1401796800">- New version 1.3.3, Update to latest ruby guidelines.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3.3-2" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Steve Traylen &lt;steve.traylen@cern.ch&gt; - 1.3.4-1" date="1402488000">- New version 1.3.4</changelog>
  <changelog author="Ha&#xEF;kel Gu&#xE9;mar &lt;hguemar@fedoraproject.org&gt; - 1.3.4-2" date="1431950400">- Fix errors with Puppet4 (patch from Lukas Bezdicka)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3.4-3" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Ga&#xEB;l Chamoulaud &lt;gchamoul@redhat.com&gt; - 1.3.4-4" date="1438257600">- Removed 0001-Fix-errors-with-Puppet-4.patch</changelog>
  <changelog author="Steve Traylen &lt;steve.traylen@cern.ch&gt; - 3.0.1-1" date="1446552000">- New version 3.0.1</changelog>
  <changelog author="Steve Traylen &lt;steve.traylen@cern.ch&gt; - 1 1.3.4-1" date="1450440000">- Revert to version 1.3.4 with an epoch of 1. rhbz#1286826</changelog>
</package>
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  <version epoch="0" ver="0.13.3" rel="13.el8"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.3-4" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.3-5" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 0.13.3-6" date="1509364800">- Fix FTBFS</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 0.13.3-7" date="1517400000">- Switch to %ldconfig_scriptlets</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.3-8" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.3-9" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.3-10" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.3-11" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.3-12" date="1580299200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
  <changelog author="Kevin Fenzi &lt;kevin@scrye.com&gt; - 0.13.3-13" date="1581768000">- Apply patch for CVE-2020-7105. Fixes bug #1796474</changelog>
</package>
<package pkgid="2d5be7758cd9a436c28f81ff91c690c7d24725311483861edad1c6f3ed04fe68" name="iksemel" arch="src">
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  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4-4" date="1326456000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4-5" date="1342699200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4-6" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4-7" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Tom&#xE1;&#x161; Mr&#xE1;z &lt;tmraz@redhat.com&gt; - 1.4-8" date="1398254400">- Rebuild for new libgcrypt</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4-9" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4-10" date="1408190400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4-11" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.4-12" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Orion Poplawski &lt;orion@cora.nwra.com&gt; - 1.5-0.1.git978b733" date="1473249600">- Update to 1.5 from git master (bug #1294251)
- Switch to openssl - does not build with gnutls 3.5 and zabbix uses openssl
- Ship python2 interface
- Drop gnutls patch fixed upstream
- Rebase sha patch
- Cleanup spec</changelog>
</package>
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  <version epoch="0" ver="1.1.1" rel="4.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 0-0.1" date="1319112000">- Initial package</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.1.1-1" date="1336564800">- Ver. 1.1.1</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.1.1-2" date="1345032000">- Add compat symlinks for old apps</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.1.1-3" date="1347451200">- Added licensing info</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.1-4" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
</package>
<package pkgid="da199e31b5661a74282e29283389c57be21fc992225aa3a2b0764d4eaa886eb7" name="jack-audio-connection-kit" arch="src">
  <version epoch="0" ver="1.9.12" rel="8.el8"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.9.10-10" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Nils Philippsen &lt;nils@tiptoe.de&gt; - 1.9.12-1" date="1514289600">- version 1.9.12
- update source URL
- update nodate, outdir, nointernalapi patches
- remove broken (undocumented, non-upstreamed) portnames patch
- remove obsolete ppc64-long/mpd, gcc* patches
- remove binary junk from README
- remove additional optimization options
- add jack_simdtests executable</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.9.12-2" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Orcan Ogetbil &lt;oget[dot]fedora[at]gmail[dot]com&gt; - 1.9.12-3" date="1518177600">- Build with RPM_LD_FLAGS exported</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.9.12-4" date="1518264000">- Switch to %ldconfig_scriptlets</changelog>
  <changelog author="Orcan Ogetbil &lt;oget[dot]fedora[at]gmail[dot]com&gt; - 1.9.12-5" date="1530792000">- Fix unversioned Python shebangs</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.9.12-6" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.9.12-7" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.9.12-8" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.9.12-9" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
</package>
<package pkgid="96165a39a372bc92a5776a5277d047571bf0f99110d4c8cc12eb5c2a6330d134" name="kronosnet-epel" arch="src">
  <version epoch="0" ver="1.28" rel="0.1.el8.1"/>
  <changelog author="Christine Caulfield &lt;ccaulfie@redhat.com&gt; - 1.18-1" date="1600862400">Rebase to 1.18
  Resolves: rhbz#1855301</changelog>
  <changelog author="Christine Caulfield &lt;ccaulfie@redhat.com&gt; - 1.22-1" date="1631793600">Rebase to 1.22
  Resolves: rhbz#1999976</changelog>
  <changelog author="Christine Caulfield &lt;ccaulfie@redhat.com&gt; - 1.24-1" date="1658923200">Rebase to 1.24
  Resolves: rhbz#2024095</changelog>
  <changelog author="Christine Caulfield &lt;ccaulfie@redhat.com&gt; - 1.24-2" date="1658923201">Don't run nozzle_up_down tests, as they don't work in RH CI
  Resolves: rhbz#2024095</changelog>
  <changelog author="Davide Cavalca &lt;dcavalca@fedoraproject.org&gt; - 1.24-2" date="1661515200">- Rename to kronosnet-epel and convert to EPEL-only package
- Remove libknet1 and libknet1-devel as they are shipped in CRB/PowerTools</changelog>
  <changelog author="Davide Cavalca &lt;dcavalca@fedoraproject.org&gt; - 1.24-3" date="1662379200">- Ensure the subpackages depend on the correct package versions for
  libknet1 and libknet1-devel
  Resolves: rhbz#2124249</changelog>
  <changelog author="Davide Cavalca &lt;dcavalca@fedoraproject.org&gt; - 1.24-4" date="1662465600">- Sync kronosnet package version with el8
  Resolves: rhbz#2124249</changelog>
  <changelog author="Davide Cavalca &lt;dcavalca@fedoraproject.org&gt; - 1.24-3.1" date="1663761600">- Sync kronosnet package version with el8
  Resolves: rhbz#2128674</changelog>
  <changelog author="Davide Cavalca &lt;dcavalca@fedoraproject.org&gt; - 1.24-4.1" date="1669291200">- Merge epel8-next into epel8 now that EL8.7 has caught up with CentOS Stream
  Resolves: rhbz#2144202</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 1.28-0.1.1" date="1742558400">- Sync kronosnet package version with c8s
- Relax dependency a bit more</changelog>
</package>
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  <version epoch="0" ver="3.100" rel="6.el8"/>
  <changelog author="RPM Fusion Release Engineering &lt;kwizart@rpmfusion.org&gt; - 3.99.5-7" date="1489924800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Yaakov Selkowitz &lt;yselkowi@redhat.com&gt; - 3.99.5-8" date="1494417600">- Rebuilt for inclusion into Fedora</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.99.5-9" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.99.5-10" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Yaakov Selkowitz &lt;yselkowi@redhat.com&gt; - 3.100-1" date="1508673600">- Update to 3.100 (#1470202, #1505172)</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.2-2" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.2-3" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.15.1b-22" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.15.1b-23" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.15.1b-24" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.15.1b-26" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.15.1b-29" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.15.1b-30" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.1-4" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.4.13-8" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.4.13-9" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.4.13-10" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.4.13-11" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 1.6.0-1" date="1518609600">- Update to upstream 1.6.0 release
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  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 1.6.0-2" date="1519041600">- Add missing BuildRequires on gcc</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.6.0-3" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.6.0-4" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.6.0-5" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.3-23" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.3-24" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.3-25" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.3-29" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.3-30" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.3-31" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-11" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-3" date="1595937600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-5" date="1611662400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
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  Related: rhbz#1991688</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.0-1" date="1675771200">- update to 2.4.0</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-11" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-1" date="1578398400">- update to 2.3.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-2" date="1580299200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-3" date="1595937600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-4" date="1602504000">- add -tools subpackage (thanks to Gerd v. Egidy)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-5" date="1611662400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-6" date="1618574400">- Rebuilt for RHEL 9 BETA on Apr 15th 2021. Related: rhbz#1947937</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.0-1" date="1675771200">- update to 2.4.0</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-2" date="1580299200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-3" date="1595937600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-4" date="1602504000">- add -tools subpackage (thanks to Gerd v. Egidy)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-5" date="1611662400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-6" date="1618574400">- Rebuilt for RHEL 9 BETA on Apr 15th 2021. Related: rhbz#1947937</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-7" date="1628510400">- Rebuilt for IMA sigs, glibc 2.34, aarch64 flags
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.0-1" date="1675771200">- update to 2.4.0</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-10" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
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  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-1" date="1578398400">- update to 2.3.0</changelog>
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  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-4" date="1602504000">- add -tools subpackage (thanks to Gerd v. Egidy)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-5" date="1611662400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
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  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-7" date="1628510400">- Rebuilt for IMA sigs, glibc 2.34, aarch64 flags
  Related: rhbz#1991688</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.5.0-1" date="1675771200">- update to 2.5.0</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-11" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-1" date="1578398400">- update to 2.3.0</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-3" date="1595937600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-4" date="1602504000">- add -tools subpackage (thanks to Gerd v. Egidy)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-5" date="1611662400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-6" date="1618574400">- Rebuilt for RHEL 9 BETA on Apr 15th 2021. Related: rhbz#1947937</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-7" date="1628510400">- Rebuilt for IMA sigs, glibc 2.34, aarch64 flags
  Related: rhbz#1991688</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.5.0-1" date="1675771200">- update to 2.5.0</changelog>
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  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0.2-9" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-10" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-11" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-12" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-13" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-14" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-15" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-16" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-17" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 2.0.0-1" date="1570708800">- Update to upstream 2.0.0 release</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-11" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-12" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-13" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-14" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-15" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-16" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0.2-17" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 2.0.0-1" date="1570708800">- Update to upstream 2.0.0 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.0.1-1" date="1630065600">- Update to upstream 2.0.1 release</changelog>
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<package pkgid="7779bc0596549a9799c215d29da9410924d76e88a48581159cddaccbad958983" name="nginx" arch="src">
  <version epoch="1" ver="1.14.1" rel="10.el8"/>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 1:1.12.1-12" date="1532001600">- add build conditional for geoip support</changelog>
  <changelog author="Lubo&#x161; Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.0-1" date="1532952000">- Resolves: #1558420 - directory permissions are now correct after processing
  USR1 signal
- Resolves: #1601414 - nginx: drop GeoIP support</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.0-2" date="1533556800">- add dependency on perl(constant)</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 1:1.14.0-3" date="1533729600">- fix PKCS#11 support (Anderson Sasaki, #1545526)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-1" date="1542628800">- new version 1.14.1
- Resolves: #1647257 - CVE-2018-16845 nginx: Denial of service and
  memory disclosure via mp4 module
- Resolves: #1647262 - CVE-2018-16844 nginx: Excessive CPU usage
  via flaw in HTTP/2 implementation
- Resolves: #1647263 - CVE-2018-16843 nginx: Excessive memory consumption
  via flaw in HTTP/2 implementation</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-2" date="1543838400">- Resolves: #1655530 - Hardening tests fail for nginx</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 1:1.14.1-3" date="1544529600">- fix unexpanded paths in nginx(8) (#1643069)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-8" date="1544616000">- enable TLS 1.3 by default (#1643647)
- TLSv1.0 and TLSv1.1 can be enabled now (#1644746)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-9" date="1567166400">- Resolves: #1744811 - CVE-2019-9511 nginx:1.14/nginx: HTTP/2: large amount of
  data request leads to denial of service
- Resolves: #1744325 - CVE-2019-9513 nginx:1.14/nginx: HTTP/2: flood using
  PRIORITY frames resulting in excessive resource consumption
- Resolves: #1745094 - CVE-2019-9516 nginx:1.14/nginx: HTTP/2: 0-length
  headers leads to denial of service</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.14.1-10" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.0
- Added http form input module version 0.12
- Added http echo module version 0.61
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
</package>
<package pkgid="01ed14975cb31349836925750afa4ac5558a612309fcaffbb2b090e0c4a3d5ff" name="nginx" arch="src">
  <version epoch="1" ver="1.14.2" rel="1.el8"/>
  <changelog author="Lubo&#x161; Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.0-1" date="1532952000">- Resolves: #1558420 - directory permissions are now correct after processing
  USR1 signal
- Resolves: #1601414 - nginx: drop GeoIP support</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.0-2" date="1533556800">- add dependency on perl(constant)</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 1:1.14.0-3" date="1533729600">- fix PKCS#11 support (Anderson Sasaki, #1545526)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-1" date="1542628800">- new version 1.14.1
- Resolves: #1647257 - CVE-2018-16845 nginx: Denial of service and
  memory disclosure via mp4 module
- Resolves: #1647262 - CVE-2018-16844 nginx: Excessive CPU usage
  via flaw in HTTP/2 implementation
- Resolves: #1647263 - CVE-2018-16843 nginx: Excessive memory consumption
  via flaw in HTTP/2 implementation</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-2" date="1543838400">- Resolves: #1655530 - Hardening tests fail for nginx</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 1:1.14.1-3" date="1544529600">- fix unexpanded paths in nginx(8) (#1643069)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-8" date="1544616000">- enable TLS 1.3 by default (#1643647)
- TLSv1.0 and TLSv1.1 can be enabled now (#1644746)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-9" date="1567166400">- Resolves: #1744811 - CVE-2019-9511 nginx:1.14/nginx: HTTP/2: large amount of
  data request leads to denial of service
- Resolves: #1744325 - CVE-2019-9513 nginx:1.14/nginx: HTTP/2: flood using
  PRIORITY frames resulting in excessive resource consumption
- Resolves: #1745094 - CVE-2019-9516 nginx:1.14/nginx: HTTP/2: 0-length
  headers leads to denial of service</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.14.1-10" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.0
- Added http form input module version 0.12
- Added http echo module version 0.61
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.14.2-10" date="1636286400">- Upgraded to NGiNX 1.14.2</changelog>
</package>
<package pkgid="a899586c3ec26624c2f13f45db9c47da089d8a8017d03d68197bad668d50b987" name="nginx" arch="src">
  <version epoch="0" ver="1.16.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.3-1" date="1485864000">- update to 1.10.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.0-1" date="1491998400">- update to 1.12.0
- added auth ldap module version 0.1
- update nginx echo module to version 0.60
- update naxsi plugin to version 0.55.3
- update rtmp module to version 1.1.11
- fixed the sticky module ng modular support
- fixed the echo module ngx_chain_t usage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.1-1" date="1504526400">- update to 1.12.1
- update nginx rtmp module to version 1.2.0
- update echo nginx module to version 0.61</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.2-1" date="1509105600">- update to 1.12.1
- update nginx push stream module to version 0.5.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-1" date="1523966400">- update to 1.14.0
- update nginx rtmp module to version 1.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-2" date="1541505600">- update naxsi plugin to version 0.56</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.1-1" date="1541505601">- update to 1.14.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.2-1" date="1553601600">- update to 1.14.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.0-1" date="1563278400">- update to 1.16.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.1-1" date="1568721600">- update to 1.16.1</changelog>
</package>
<package pkgid="812c5517c8a836a2116ffa56ca7c4ba523f419fb6c1814a9bd6f003c9bdcf77e" name="nginx" arch="src">
  <version epoch="1" ver="1.16.1" rel="2.el8"/>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 1:1.14.0-3" date="1533729600">- fix PKCS#11 support (Anderson Sasaki, #1545526)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-1" date="1542628800">- new version 1.14.1
- Resolves: #1647257 - CVE-2018-16845 nginx: Denial of service and
  memory disclosure via mp4 module
- Resolves: #1647262 - CVE-2018-16844 nginx: Excessive CPU usage
  via flaw in HTTP/2 implementation
- Resolves: #1647263 - CVE-2018-16843 nginx: Excessive memory consumption
  via flaw in HTTP/2 implementation</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-2" date="1543838400">- Resolves: #1655530 - Hardening tests fail for nginx</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 1:1.14.1-3" date="1544529600">- fix unexpanded paths in nginx(8) (#1643069)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-8" date="1544616000">- enable TLS 1.3 by default (#1643647)
- TLSv1.0 and TLSv1.1 can be enabled now (#1644746)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-1" date="1557144000">- new version 1.16.0
- enable ngx_stream_ssl_preread module
- main package does NOT require all-modules package</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-2" date="1561550400">- Resolves: #1718929 - ssl_protocols config option has faulty behavior
  in nginx:1.16</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.1-1" date="1567080000">- update to 1.16.1
- Resolves: #1745697 - CVE-2019-9511 nginx:1.16/nginx: HTTP/2: large amount
  of data request leads to denial of service
- Resolves: #1745690 - CVE-2019-9513 nginx:1.16/nginx: HTTP/2: flood using
  PRIORITY frames resulting in excessive resource consumption
- Resolves: #1745645 - CVE-2019-9516 nginx:1.16/nginx: HTTP/2: 0-length
  headers leads to denial of service</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.1-1.1" date="1606219200">- Resolves: #1898952 - CVE 2019-20372 nginx:1.16/nginx: HTTP request smuggling
  via error pages in http/ngx_http_special_response.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.16.1-2" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.0
- Added http form input module version 0.12
- Added http echo module version 0.61
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
</package>
<package pkgid="b025ac134fa65bbbe9e4401233abe7cf5bcfb54e11ddbcad772b3380790f1fbc" name="nginx" arch="src">
  <version epoch="1" ver="1.16.1" rel="3.el8"/>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-1" date="1542628800">- new version 1.14.1
- Resolves: #1647257 - CVE-2018-16845 nginx: Denial of service and
  memory disclosure via mp4 module
- Resolves: #1647262 - CVE-2018-16844 nginx: Excessive CPU usage
  via flaw in HTTP/2 implementation
- Resolves: #1647263 - CVE-2018-16843 nginx: Excessive memory consumption
  via flaw in HTTP/2 implementation</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-2" date="1543838400">- Resolves: #1655530 - Hardening tests fail for nginx</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 1:1.14.1-3" date="1544529600">- fix unexpanded paths in nginx(8) (#1643069)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-8" date="1544616000">- enable TLS 1.3 by default (#1643647)
- TLSv1.0 and TLSv1.1 can be enabled now (#1644746)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-1" date="1557144000">- new version 1.16.0
- enable ngx_stream_ssl_preread module
- main package does NOT require all-modules package</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-2" date="1561550400">- Resolves: #1718929 - ssl_protocols config option has faulty behavior
  in nginx:1.16</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.1-1" date="1567080000">- update to 1.16.1
- Resolves: #1745697 - CVE-2019-9511 nginx:1.16/nginx: HTTP/2: large amount
  of data request leads to denial of service
- Resolves: #1745690 - CVE-2019-9513 nginx:1.16/nginx: HTTP/2: flood using
  PRIORITY frames resulting in excessive resource consumption
- Resolves: #1745645 - CVE-2019-9516 nginx:1.16/nginx: HTTP/2: 0-length
  headers leads to denial of service</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.1-1.1" date="1606219200">- Resolves: #1898952 - CVE 2019-20372 nginx:1.16/nginx: HTTP request smuggling
  via error pages in http/ngx_http_special_response.c</changelog>
  <changelog author="Lubo&#x161; Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.1-2" date="1621944000">- Resolves: #1963174 - CVE-2021-23017 nginx:1.16/nginx: Off-by-one in
  ngx_resolver_copy() when labels are followed by a pointer to a root
  domain name</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.16.1-3" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.0
- Added http form input module version 0.12
- Added http echo module version 0.61
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
</package>
<package pkgid="20a6967e1b460eccdaff9ff9da5cb69d3a9cb84db32711affd7da3b68b141f6f" name="nginx" arch="src">
  <version epoch="1" ver="1.18.0" rel="4.el8"/>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-2" date="1543838400">- Resolves: #1655530 - Hardening tests fail for nginx</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 1:1.14.1-3" date="1544529600">- fix unexpanded paths in nginx(8) (#1643069)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-8" date="1544616000">- enable TLS 1.3 by default (#1643647)
- TLSv1.0 and TLSv1.1 can be enabled now (#1644746)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-1" date="1557144000">- new version 1.16.0
- enable ngx_stream_ssl_preread module
- main package does NOT require all-modules package</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-2" date="1561550400">- Resolves: #1718929 - ssl_protocols config option has faulty behavior
  in nginx:1.16</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.1-1" date="1567080000">- update to 1.16.1
- Resolves: #1745697 - CVE-2019-9511 nginx:1.16/nginx: HTTP/2: large amount
  of data request leads to denial of service
- Resolves: #1745690 - CVE-2019-9513 nginx:1.16/nginx: HTTP/2: flood using
  PRIORITY frames resulting in excessive resource consumption
- Resolves: #1745645 - CVE-2019-9516 nginx:1.16/nginx: HTTP/2: 0-length
  headers leads to denial of service</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-2" date="1587556800">- new version 1.18.0
- Resolves: #1668717 - [RFE] Support loading certificates from hardware token
  (PKCS#11)
- Increased types_hash_max_size to 4096 in default config
- Drop location / from default config (rhbz#1564768)
- Drop default_sever from default config (rhbz#1373822)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3" date="1605182400">- Resolves: #1651377 - centralizing default index.html on nginx
- Resolves: #1825683 - Outdated Red Hat branding used in nginx default pages</changelog>
  <changelog author="Lubo&#x161; Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3.1" date="1621944000">- Resolves: #1963178 - CVE-2021-23017 nginx:1.18/nginx: Off-by-one in
  ngx_resolver_copy() when labels are followed by a pointer to a root
  domain name</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.18.0-4" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.1
- Added http form input module version 0.12
- Added http echo module version 0.62
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
</package>
<package pkgid="5031472c31a27a8e08cb7835bda1d04f2752d7d3d7c5520fff2575a19c4be00a" name="nginx" arch="src">
  <version epoch="1" ver="1.20.0" rel="1.el8"/>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 1:1.14.1-3" date="1544529600">- fix unexpanded paths in nginx(8) (#1643069)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-8" date="1544616000">- enable TLS 1.3 by default (#1643647)
- TLSv1.0 and TLSv1.1 can be enabled now (#1644746)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-1" date="1557144000">- new version 1.16.0
- enable ngx_stream_ssl_preread module
- main package does NOT require all-modules package</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-2" date="1561550400">- Resolves: #1718929 - ssl_protocols config option has faulty behavior
  in nginx:1.16</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.1-1" date="1567080000">- update to 1.16.1
- Resolves: #1745697 - CVE-2019-9511 nginx:1.16/nginx: HTTP/2: large amount
  of data request leads to denial of service
- Resolves: #1745690 - CVE-2019-9513 nginx:1.16/nginx: HTTP/2: flood using
  PRIORITY frames resulting in excessive resource consumption
- Resolves: #1745645 - CVE-2019-9516 nginx:1.16/nginx: HTTP/2: 0-length
  headers leads to denial of service</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-2" date="1587556800">- new version 1.18.0
- Resolves: #1668717 - [RFE] Support loading certificates from hardware token
  (PKCS#11)
- Increased types_hash_max_size to 4096 in default config
- Drop location / from default config (rhbz#1564768)
- Drop default_sever from default config (rhbz#1373822)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3" date="1605182400">- Resolves: #1651377 - centralizing default index.html on nginx
- Resolves: #1825683 - Outdated Red Hat branding used in nginx default pages</changelog>
  <changelog author="Lubo&#x161; Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3.1" date="1621944000">- Resolves: #1963178 - CVE-2021-23017 nginx:1.18/nginx: Off-by-one in
  ngx_resolver_copy() when labels are followed by a pointer to a root
  domain name</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.18.0-4" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.1
- Added http form input module version 0.12
- Added http echo module version 0.62
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-0-1" date="1636372800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
</package>
<package pkgid="8a8739c5dd4084f86efa71ccea6550b3a824a4a1be09c6df1a7400dcdb652597" name="nginx" arch="src">
  <version epoch="1" ver="1.20.1" rel="1.el8"/>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.14.1-8" date="1544616000">- enable TLS 1.3 by default (#1643647)
- TLSv1.0 and TLSv1.1 can be enabled now (#1644746)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-1" date="1557144000">- new version 1.16.0
- enable ngx_stream_ssl_preread module
- main package does NOT require all-modules package</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-2" date="1561550400">- Resolves: #1718929 - ssl_protocols config option has faulty behavior
  in nginx:1.16</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.1-1" date="1567080000">- update to 1.16.1
- Resolves: #1745697 - CVE-2019-9511 nginx:1.16/nginx: HTTP/2: large amount
  of data request leads to denial of service
- Resolves: #1745690 - CVE-2019-9513 nginx:1.16/nginx: HTTP/2: flood using
  PRIORITY frames resulting in excessive resource consumption
- Resolves: #1745645 - CVE-2019-9516 nginx:1.16/nginx: HTTP/2: 0-length
  headers leads to denial of service</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-2" date="1587556800">- new version 1.18.0
- Resolves: #1668717 - [RFE] Support loading certificates from hardware token
  (PKCS#11)
- Increased types_hash_max_size to 4096 in default config
- Drop location / from default config (rhbz#1564768)
- Drop default_sever from default config (rhbz#1373822)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3" date="1605182400">- Resolves: #1651377 - centralizing default index.html on nginx
- Resolves: #1825683 - Outdated Red Hat branding used in nginx default pages</changelog>
  <changelog author="Lubo&#x161; Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3.1" date="1621944000">- Resolves: #1963178 - CVE-2021-23017 nginx:1.18/nginx: Off-by-one in
  ngx_resolver_copy() when labels are followed by a pointer to a root
  domain name</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.18.0-4" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.1
- Added http form input module version 0.12
- Added http echo module version 0.62
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-0-1" date="1636372800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372801">- update to 1.20.1</changelog>
</package>
<package pkgid="e6e2ed0a0f06d84c6ffea19c11fa6cc62fd83c1b1ab94872333b5dc01866af87" name="nginx" arch="src">
  <version epoch="1" ver="1.20.1" rel="2.el8"/>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-1" date="1557144000">- new version 1.16.0
- enable ngx_stream_ssl_preread module
- main package does NOT require all-modules package</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-2" date="1561550400">- Resolves: #1718929 - ssl_protocols config option has faulty behavior
  in nginx:1.16</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.1-1" date="1567080000">- update to 1.16.1
- Resolves: #1745697 - CVE-2019-9511 nginx:1.16/nginx: HTTP/2: large amount
  of data request leads to denial of service
- Resolves: #1745690 - CVE-2019-9513 nginx:1.16/nginx: HTTP/2: flood using
  PRIORITY frames resulting in excessive resource consumption
- Resolves: #1745645 - CVE-2019-9516 nginx:1.16/nginx: HTTP/2: 0-length
  headers leads to denial of service</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-2" date="1587556800">- new version 1.18.0
- Resolves: #1668717 - [RFE] Support loading certificates from hardware token
  (PKCS#11)
- Increased types_hash_max_size to 4096 in default config
- Drop location / from default config (rhbz#1564768)
- Drop default_sever from default config (rhbz#1373822)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3" date="1605182400">- Resolves: #1651377 - centralizing default index.html on nginx
- Resolves: #1825683 - Outdated Red Hat branding used in nginx default pages</changelog>
  <changelog author="Lubo&#x161; Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3.1" date="1621944000">- Resolves: #1963178 - CVE-2021-23017 nginx:1.18/nginx: Off-by-one in
  ngx_resolver_copy() when labels are followed by a pointer to a root
  domain name</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.18.0-4" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.1
- Added http form input module version 0.12
- Added http echo module version 0.62
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-0-1" date="1636372800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372801">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
</package>
<package pkgid="577500e429215dc4116a82cdbfad7189e04ac885a56761c163c6ab2177095f47" name="nginx" arch="src">
  <version epoch="1" ver="1.20.2" rel="1.el8"/>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.0-2" date="1561550400">- Resolves: #1718929 - ssl_protocols config option has faulty behavior
  in nginx:1.16</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.1-1" date="1567080000">- update to 1.16.1
- Resolves: #1745697 - CVE-2019-9511 nginx:1.16/nginx: HTTP/2: large amount
  of data request leads to denial of service
- Resolves: #1745690 - CVE-2019-9513 nginx:1.16/nginx: HTTP/2: flood using
  PRIORITY frames resulting in excessive resource consumption
- Resolves: #1745645 - CVE-2019-9516 nginx:1.16/nginx: HTTP/2: 0-length
  headers leads to denial of service</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-2" date="1587556800">- new version 1.18.0
- Resolves: #1668717 - [RFE] Support loading certificates from hardware token
  (PKCS#11)
- Increased types_hash_max_size to 4096 in default config
- Drop location / from default config (rhbz#1564768)
- Drop default_sever from default config (rhbz#1373822)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3" date="1605182400">- Resolves: #1651377 - centralizing default index.html on nginx
- Resolves: #1825683 - Outdated Red Hat branding used in nginx default pages</changelog>
  <changelog author="Lubo&#x161; Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3.1" date="1621944000">- Resolves: #1963178 - CVE-2021-23017 nginx:1.18/nginx: Off-by-one in
  ngx_resolver_copy() when labels are followed by a pointer to a root
  domain name</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.18.0-4" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.1
- Added http form input module version 0.12
- Added http echo module version 0.62
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-0-1" date="1636372800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372801">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
</package>
<package pkgid="76f0d9801b7666a9a3ae84b6e4df2206c36315793decf9b015c103da8f156366" name="nginx" arch="src">
  <version epoch="1" ver="1.20.2" rel="2.el8"/>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.16.1-1" date="1567080000">- update to 1.16.1
- Resolves: #1745697 - CVE-2019-9511 nginx:1.16/nginx: HTTP/2: large amount
  of data request leads to denial of service
- Resolves: #1745690 - CVE-2019-9513 nginx:1.16/nginx: HTTP/2: flood using
  PRIORITY frames resulting in excessive resource consumption
- Resolves: #1745645 - CVE-2019-9516 nginx:1.16/nginx: HTTP/2: 0-length
  headers leads to denial of service</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-2" date="1587556800">- new version 1.18.0
- Resolves: #1668717 - [RFE] Support loading certificates from hardware token
  (PKCS#11)
- Increased types_hash_max_size to 4096 in default config
- Drop location / from default config (rhbz#1564768)
- Drop default_sever from default config (rhbz#1373822)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3" date="1605182400">- Resolves: #1651377 - centralizing default index.html on nginx
- Resolves: #1825683 - Outdated Red Hat branding used in nginx default pages</changelog>
  <changelog author="Lubo&#x161; Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3.1" date="1621944000">- Resolves: #1963178 - CVE-2021-23017 nginx:1.18/nginx: Off-by-one in
  ngx_resolver_copy() when labels are followed by a pointer to a root
  domain name</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.18.0-4" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.1
- Added http form input module version 0.12
- Added http echo module version 0.62
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-0-1" date="1636372800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372801">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
</package>
<package pkgid="3dd6eff271b7cadabe8df55aba79490b32139599562e69dd84c218bebcb11e7a" name="nginx" arch="src">
  <version epoch="1" ver="1.22.0" rel="1.el8"/>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-2" date="1587556800">- new version 1.18.0
- Resolves: #1668717 - [RFE] Support loading certificates from hardware token
  (PKCS#11)
- Increased types_hash_max_size to 4096 in default config
- Drop location / from default config (rhbz#1564768)
- Drop default_sever from default config (rhbz#1373822)</changelog>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3" date="1605182400">- Resolves: #1651377 - centralizing default index.html on nginx
- Resolves: #1825683 - Outdated Red Hat branding used in nginx default pages</changelog>
  <changelog author="Lubo&#x161; Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3.1" date="1621944000">- Resolves: #1963178 - CVE-2021-23017 nginx:1.18/nginx: Off-by-one in
  ngx_resolver_copy() when labels are followed by a pointer to a root
  domain name</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.18.0-4" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.1
- Added http form input module version 0.12
- Added http echo module version 0.62
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-0-1" date="1636372800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372801">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
</package>
<package pkgid="cfd607c4ab632b43772e454b2f56dd6b5fddedb77c23972bd7480c0cbeae125b" name="nginx" arch="src">
  <version epoch="1" ver="1.22.0" rel="2.el8"/>
  <changelog author="Lubos Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3" date="1605182400">- Resolves: #1651377 - centralizing default index.html on nginx
- Resolves: #1825683 - Outdated Red Hat branding used in nginx default pages</changelog>
  <changelog author="Lubo&#x161; Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3.1" date="1621944000">- Resolves: #1963178 - CVE-2021-23017 nginx:1.18/nginx: Off-by-one in
  ngx_resolver_copy() when labels are followed by a pointer to a root
  domain name</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.18.0-4" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.1
- Added http form input module version 0.12
- Added http echo module version 0.62
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-0-1" date="1636372800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372801">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
</package>
<package pkgid="c0c1afc7af253840b03b96655a9c1a13d354e92bd049e245559dc1e2f2be4440" name="nginx" arch="src">
  <version epoch="1" ver="1.22.1" rel="1.el8"/>
  <changelog author="Lubo&#x161; Uhliarik &lt;luhliari@redhat.com&gt; - 1:1.18.0-3.1" date="1621944000">- Resolves: #1963178 - CVE-2021-23017 nginx:1.18/nginx: Off-by-one in
  ngx_resolver_copy() when labels are followed by a pointer to a root
  domain name</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.18.0-4" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.1
- Added http form input module version 0.12
- Added http echo module version 0.62
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-0-1" date="1636372800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372801">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
</package>
<package pkgid="b691ad7cb66b6d3227bed2b95a2d8548d9966e61be6684ac5877429919580059" name="nginx" arch="src">
  <version epoch="1" ver="1.24.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.18.0-4" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.1
- Added http form input module version 0.12
- Added http echo module version 0.62
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-0-1" date="1636372800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372801">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
</package>
<package pkgid="c8b94b4f6598f9fda2f28fdc1f2187216c74ddeecd2f189423354fee6094e88e" name="nginx" arch="src">
  <version epoch="1" ver="1.26.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-0-1" date="1636372800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372801">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
</package>
<package pkgid="3c48b2081f97c69499e6c2364af9e8dfabd30d16e885c302e218af68fe483162" name="nginx" arch="src">
  <version epoch="1" ver="1.26.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372800">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
</package>
<package pkgid="42049dbd05d2968859adc29707b4e34645c64cfbf9e821411263a30874d5ca36" name="nginx" arch="src">
  <version epoch="1" ver="1.26.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
</package>
<package pkgid="6fe5bbedc5baf8acfb8f02882a947d21c207ef289c07584934a2993e32b50e07" name="nginx" arch="src">
  <version epoch="1" ver="1.26.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.3-1" date="1738756800">- update to 1.26.3</changelog>
</package>
<package pkgid="8f5910bc3d2688098e57246bf42fd03dfdd4a68dc47dd174dfedeeb0c9f3855c" name="nginx" arch="src">
  <version epoch="1" ver="1.28.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283200">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.3-1" date="1738756800">- update to 1.26.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.0-1" date="1745409600">- update to 1.28.0</changelog>
</package>
<package pkgid="fc395b2be19d8b11f18f7365050f1cba2f8389eef2165d4f2183eb4e0755227d" name="nginx" arch="src">
  <version epoch="1" ver="1.28.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.3-1" date="1738756800">- update to 1.26.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.0-1" date="1745409600">- update to 1.28.0</changelog>
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- update njs module to version 0.9.6</changelog>
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- update njs module to version 0.9.6</changelog>
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- update njs module to version 0.9.6</changelog>
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- update http-proxy-connect-module to version 0.0.7</changelog>
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- update njs module to version 0.9.6</changelog>
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- update http-proxy-connect-module to version 0.0.7</changelog>
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- update njs module to version 0.9.6</changelog>
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- update http-proxy-connect-module to version 0.0.7</changelog>
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- update njs module to version 0.9.9</changelog>
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- update njs module to version 0.9.6</changelog>
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- update http-proxy-connect-module to version 0.0.7</changelog>
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- update njs module to version 0.9.9</changelog>
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- update njs module to version 0.9.6</changelog>
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- update http-proxy-connect-module to version 0.0.7</changelog>
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- update njs module to version 0.9.9</changelog>
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  <changelog author="Fabio Alessandro Locati &lt;fale@fedoraproject.org&gt; - 1.47-0.1.a39256b7" date="1477742400">- Rebase on current master branch
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.47-0.2.a39256b" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.47-0.4.a39256b" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.47-0.5.a39256b" date="1518091200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.47-0.6.a39256b" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.47-3.1.20180101git9b59468" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
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  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.3-1" date="1336910400">- Update to 0.1.3</changelog>
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  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.1.3-4" date="1409486400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 0.1.5-1" date="1490011200">- Update to 0.1.5</changelog>
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  <changelog author="RPM Fusion Release Engineering &lt;leigh123linux@googlemail.com&gt; - 0.1.5-3" date="1519905600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="RPM Fusion Release Engineering &lt;leigh123linux@gmail.com&gt; - 0.1.5-4" date="1532692800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 0.1.5-5" date="1534680000">- Rebuilt for Fedora 29 Mass Rebuild binutils issue</changelog>
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  <changelog author="RPM Fusion Release Engineering &lt;leigh123linux@googlemail.com&gt; - 0.1.5-3" date="1519905600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="RPM Fusion Release Engineering &lt;leigh123linux@gmail.com&gt; - 0.1.5-4" date="1532692800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 0.1.5-5" date="1534680000">- Rebuilt for Fedora 29 Mass Rebuild binutils issue</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.5-6" date="1544529600">- Spec file clean-up</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.1.6-1" date="1659355200">- Update to 0.1.6</changelog>
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  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
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- New packages: compression, emergency, fraud_detection,
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  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="b2de6bde571c9cdb1a799142cd821e55853698647e8f3bf1a13be19248919227" name="opensips" arch="src">
  <version epoch="0" ver="3.1.1" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="fc201f854f1b79dac125e1167c6ab6750a1a179bbead6346df22a68942c9050b" name="opensips" arch="src">
  <version epoch="0" ver="3.1.10" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="8eaf9679cdb29656d0f7bccfd1fe48316789f3ddaadf12c063c5134759fbd112" name="opensips" arch="src">
  <version epoch="0" ver="3.1.11" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="88f77857167d156cb1fade2e21da3264ab3093c88ec57fdfeae23faf3452eaeb" name="opensips" arch="src">
  <version epoch="0" ver="3.1.12" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="7b4054d255c03d82551163f2961ccf56eba0614bb44660c74cf2f9d325d03d31" name="opensips" arch="src">
  <version epoch="0" ver="3.1.13" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="51bb2316f72f69110a94323742701fd5793aee44928ad05fc9f677b96df34e93" name="opensips" arch="src">
  <version epoch="0" ver="3.1.14" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="630dad7d906d672f1e34501618a99b5ff0264580acae30397352490e3ebdf1c2" name="opensips" arch="src">
  <version epoch="0" ver="3.1.15" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="d71e18cfa96034d24950fc42e7e12577488d117b515ef67a6a7d86c2df64f5d8" name="opensips" arch="src">
  <version epoch="0" ver="3.1.16" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.1.16-1" date="1687348800">- OpenSIPS minor stable release: 3.1.16-1</changelog>
</package>
<package pkgid="73d07067002b30738437d425fa26f25975adb550408477757625e504d5714af5" name="opensips" arch="src">
  <version epoch="0" ver="3.1.17" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.1.16-1" date="1687348800">- OpenSIPS minor stable release: 3.1.16-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.1.17-1" date="1693483200">- OpenSIPS minor stable release: 3.1.17-1</changelog>
</package>
<package pkgid="8596052311cd9bd61a21f3cb37c6f3f81eda54f09302cd67c933960c0b33320b" name="opensips" arch="src">
  <version epoch="0" ver="3.1.2" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="ff352b127fde541070667b09853345937d0e7d53647b542144aacc0e230f897b" name="opensips" arch="src">
  <version epoch="0" ver="3.1.3" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="2baed8c376537cd43130a70bc93a70e93bb81f11cf6aea083ae2505de09f7b9b" name="opensips" arch="src">
  <version epoch="0" ver="3.1.4" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="5c76777384c3e9348a7871ee146e178627b64cc1e78bf1b20e05b11f129a1d77" name="opensips" arch="src">
  <version epoch="0" ver="3.1.5" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="ff68b606b50e46a609de1145f71416aa8d9b13b2aefa7aa830a5b35fb68c937e" name="opensips" arch="src">
  <version epoch="0" ver="3.1.6" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="69bad71be9d099f4cd905c08380a0c17b8f26fa9f67684e816931645a5674168" name="opensips" arch="src">
  <version epoch="0" ver="3.1.7" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="f5a0c3c006004835edf69035c1603c1a1796fa21dc0ca28fc89eebeeed59366d" name="opensips" arch="src">
  <version epoch="0" ver="3.1.8" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="63054221ddf1854b31bef7f728487165aff7b801aca1ffee99d141b69f2e8924" name="opensips" arch="src">
  <version epoch="0" ver="3.1.9" rel="1.el8"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="94b5d21df93277b3fcf03d7c83651f9f16b5ae3b417bb11a7b2d2e4edf029ef9" name="opensips" arch="src">
  <version epoch="0" ver="3.2.0" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="e8e38b64e38bdec5ffaca9f316943bd16f1d440bfea5df363f738f25382f8e62" name="opensips" arch="src">
  <version epoch="0" ver="3.2.1" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="11082e019171080062c006a036b4722335ed98498ad67209623e591a1ef0ad26" name="opensips" arch="src">
  <version epoch="0" ver="3.2.10" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="be167a8402426310e34930093403a8eb767348cd768af9d3fc07e4a64783f085" name="opensips" arch="src">
  <version epoch="0" ver="3.2.11" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="1cf441419e8b0fbf3cef76fda07d467c79df67b8d5f1e30847176117bf80003f" name="opensips" arch="src">
  <version epoch="0" ver="3.2.12" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="97a3adb3363cb4e0de7a48e96647728c7b6b93bb67b6ef002b71b2de072ccca9" name="opensips" arch="src">
  <version epoch="0" ver="3.2.13" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
</package>
<package pkgid="d7921eba75240d8788afc2971d051aeb7a2218b080f08f4ff761f7e7866d5f6f" name="opensips" arch="src">
  <version epoch="0" ver="3.2.14" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.14-1" date="1693483200">- OpenSIPS minor stable release: 3.2.14-1</changelog>
</package>
<package pkgid="055d8caa47f6c2162767010b79bbe9ecbb09e14d9fe1e18c1ee04981f518375a" name="opensips" arch="src">
  <version epoch="0" ver="3.2.15" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.14-1" date="1693483200">- OpenSIPS minor stable release: 3.2.14-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.15-1" date="1697630400">- OpenSIPS minor stable release: 3.2.15-1</changelog>
</package>
<package pkgid="bd7793000bb209219a5974237ab128bfb5d588a30776efe040e599db0434a2fa" name="opensips" arch="src">
  <version epoch="0" ver="3.2.16" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.14-1" date="1693483200">- OpenSIPS minor stable release: 3.2.14-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.15-1" date="1697630400">- OpenSIPS minor stable release: 3.2.15-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.16-1" date="1703073600">- OpenSIPS minor stable release: 3.2.16-1</changelog>
</package>
<package pkgid="793267fd56ff09978ed4f9dad1bc23b65a3f017bf60eb8215f1d6e91778d2715" name="opensips" arch="src">
  <version epoch="0" ver="3.2.17" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.14-1" date="1693483200">- OpenSIPS minor stable release: 3.2.14-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.15-1" date="1697630400">- OpenSIPS minor stable release: 3.2.15-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.16-1" date="1703073600">- OpenSIPS minor stable release: 3.2.16-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.17-1" date="1708516800">- OpenSIPS minor stable release: 3.2.17-1</changelog>
</package>
<package pkgid="15418d48381ab6ee5624147f9a0b33f85f1a075f0a35cdd58be144f1fe5fc2a9" name="opensips" arch="src">
  <version epoch="0" ver="3.2.18" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.14-1" date="1693483200">- OpenSIPS minor stable release: 3.2.14-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.15-1" date="1697630400">- OpenSIPS minor stable release: 3.2.15-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.16-1" date="1703073600">- OpenSIPS minor stable release: 3.2.16-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.17-1" date="1708516800">- OpenSIPS minor stable release: 3.2.17-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.18-1" date="1713441600">- OpenSIPS minor stable release: 3.2.18-1</changelog>
</package>
<package pkgid="1f4433414205e1450595d3c3969ed3de7662c48c71cc1c08756e040fc56a6158" name="opensips" arch="src">
  <version epoch="0" ver="3.2.19" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.14-1" date="1693483200">- OpenSIPS minor stable release: 3.2.14-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.15-1" date="1697630400">- OpenSIPS minor stable release: 3.2.15-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.16-1" date="1703073600">- OpenSIPS minor stable release: 3.2.16-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.17-1" date="1708516800">- OpenSIPS minor stable release: 3.2.17-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.18-1" date="1713441600">- OpenSIPS minor stable release: 3.2.18-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.19-1" date="1718798400">- OpenSIPS minor stable release: 3.2.19-1</changelog>
</package>
<package pkgid="e3f4f9687b61654040bd1308287b5ec61d60d23f412bfe5f055579882c7b28b7" name="opensips" arch="src">
  <version epoch="0" ver="3.2.2" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="e4c2611160b05a192adf376d8a61dcdd671676aef9566650a27249496416b8d4" name="opensips" arch="src">
  <version epoch="0" ver="3.2.3" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="6d86f82df5c752dee3a0e0528be84e5909a67c2ae0639f00f996d461bd22637d" name="opensips" arch="src">
  <version epoch="0" ver="3.2.4" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="9b0b3839f6409124a898019de29cf991e81c170960f93ee558c387507b393a98" name="opensips" arch="src">
  <version epoch="0" ver="3.2.5" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="2889c982a8ca95d6dcac0127212d83a20ca1ce719dff391a2fa6c04a9e7e4bae" name="opensips" arch="src">
  <version epoch="0" ver="3.2.6" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="bb69d20a3321a255f016657335613db3592bc96f27e749589b522515bd0a6ae0" name="opensips" arch="src">
  <version epoch="0" ver="3.2.7" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="814f7aea8e874cf82e9b6c330a487302db775337d5965fa2db3a722f3697d904" name="opensips" arch="src">
  <version epoch="0" ver="3.2.8" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="56f68b2ba0a4d374b480013e343d49d322b9f193aae424449072db98fac7befa" name="opensips" arch="src">
  <version epoch="0" ver="3.2.9" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="faec2b83b0195c45764e4f689ce7c4a0f27cdbab93db2640d49c0bd347e9e354" name="opensips" arch="src">
  <version epoch="0" ver="3.3.0" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="6ed3bcbb89c77d5d0d6ad805703e6e8633621d42d43321127dcce6d93cd46824" name="opensips" arch="src">
  <version epoch="0" ver="3.3.1" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="242715ba74c9494b5ea2946714ec58238984f531313ae5adf243a3fcfe00fe88" name="opensips" arch="src">
  <version epoch="0" ver="3.3.10" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.6-1" date="1687348800">- OpenSIPS minor stable release: 3.3.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.7-1" date="1693483200">- OpenSIPS minor stable release: 3.3.7-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.8-1" date="1697630400">- OpenSIPS minor stable release: 3.3.8-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.9-1" date="1703073600">- OpenSIPS minor stable release: 3.3.9-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.10-1" date="1708516800">- OpenSIPS minor stable release: 3.3.10-1</changelog>
</package>
<package pkgid="debbd067a758ff855edc13c12e62c6afcda1a024e44690e0f540d2a089ea428c" name="opensips" arch="src">
  <version epoch="0" ver="3.3.2" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="31af8f0d921e754e77764ab0844d9171621fd6af9f5ebb89ec46f6145a5b4fde" name="opensips" arch="src">
  <version epoch="0" ver="3.3.3" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="11eaace7ee2d59f29cd162b89fa44a46039905b71c51a622608ff18e8e58bd52" name="opensips" arch="src">
  <version epoch="0" ver="3.3.4" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="c6d1c25623db924f59c632ce175d4b53da176266b3aa28800bc4c4131d974fdb" name="opensips" arch="src">
  <version epoch="0" ver="3.3.5" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="5eb7ab8f9177556f2f63e254a5ee99d8fca664f341d34b5b58af61b59bac1e83" name="opensips" arch="src">
  <version epoch="0" ver="3.3.6" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.6-1" date="1687348800">- OpenSIPS minor stable release: 3.3.6-1</changelog>
</package>
<package pkgid="8581a40512793d6e77fd107b935ef69adbbfa48ef00c0bfab7ef292f5fb8558b" name="opensips" arch="src">
  <version epoch="0" ver="3.3.7" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.6-1" date="1687348800">- OpenSIPS minor stable release: 3.3.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.7-1" date="1693483200">- OpenSIPS minor stable release: 3.3.7-1</changelog>
</package>
<package pkgid="265f2ab741fad8160f2ab6ebcbecc1683f11eb8bbeca6c246d2097aadb6789b2" name="opensips" arch="src">
  <version epoch="0" ver="3.3.8" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.6-1" date="1687348800">- OpenSIPS minor stable release: 3.3.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.7-1" date="1693483200">- OpenSIPS minor stable release: 3.3.7-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.8-1" date="1697630400">- OpenSIPS minor stable release: 3.3.8-1</changelog>
</package>
<package pkgid="7d559dc0ba1e1bf06ba27dc67d1995b43dcf2c512df4535790fb37a6bda37efd" name="opensips" arch="src">
  <version epoch="0" ver="3.3.9" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.6-1" date="1687348800">- OpenSIPS minor stable release: 3.3.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.7-1" date="1693483200">- OpenSIPS minor stable release: 3.3.7-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.8-1" date="1697630400">- OpenSIPS minor stable release: 3.3.8-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.9-1" date="1703073600">- OpenSIPS minor stable release: 3.3.9-1</changelog>
</package>
<package pkgid="dae4fc2384aad6eb858aef73127fe09120441382d6e67198ec7c58d4017ae50b" name="opensips" arch="src">
  <version epoch="0" ver="3.4.0.beta" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="feae82a36b7a14773c58613c5ca86a72ad51152c0153c8e3ff01252c8c5a99b7" name="opensips" arch="src">
  <version epoch="0" ver="3.4.0.rc1" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
</package>
<package pkgid="02eff8dcecd9882a7e42343f71504c255db96fcd50a25846273d21684bc42784" name="opensips" arch="src">
  <version epoch="0" ver="3.4.1" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-1" date="1689768000">- OpenSIPS 3.4.0 stable release: 3.4.0-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
</package>
<package pkgid="8bd5252989be7ff6c6c4b5ce2fcac0e3f0752f617c3bbf56a2d45f743d944437" name="opensips" arch="src">
  <version epoch="0" ver="3.4.10" rel="1.el8"/>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.4-1" date="1708516800">- OpenSIPS minor stable release: 3.4.4-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.5-1" date="1713441600">- OpenSIPS minor stable release: 3.4.5-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.6-1" date="1718798400">- OpenSIPS minor stable release: 3.4.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.7-1" date="1721822400">- OpenSIPS minor stable release: 3.4.7-1</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.4.7-1" date="1724068800">- Replace deprecated dependency for radius modules</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.4.7-1" date="1724155200">- Add python-setuptools dependency</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.8-1" date="1724241600">- OpenSIPS minor stable release: 3.4.8-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.9-1" date="1729684800">- OpenSIPS minor stable release: 3.4.9-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.10-1" date="1734523200">- OpenSIPS minor stable release: 3.4.10-1</changelog>
</package>
<package pkgid="294f50060c494837d46e7adda63c075f067983df71b9f7309ff078d8bc8ce3cc" name="opensips" arch="src">
  <version epoch="0" ver="3.4.2" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-1" date="1689768000">- OpenSIPS 3.4.0 stable release: 3.4.0-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
</package>
<package pkgid="59f001417326a59a278c0dd8a7c06ffa316bcc7e7b32dc2762e249cd3dc0bc29" name="opensips" arch="src">
  <version epoch="0" ver="3.4.3" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-1" date="1689768000">- OpenSIPS 3.4.0 stable release: 3.4.0-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
</package>
<package pkgid="f222981d43ce95d552b202cb0445f7179da9042c03a48009230c224f8c0f5648" name="opensips" arch="src">
  <version epoch="0" ver="3.4.4" rel="1.el8"/>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-1" date="1689768000">- OpenSIPS 3.4.0 stable release: 3.4.0-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.4-1" date="1708516800">- OpenSIPS minor stable release: 3.4.4-1</changelog>
</package>
<package pkgid="62c27b3e2ceced73d08d034883339b625d3a2f9606a0a2c6efe3d16bf6d2b54a" name="opensips" arch="src">
  <version epoch="0" ver="3.4.5" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-1" date="1689768000">- OpenSIPS 3.4.0 stable release: 3.4.0-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.4-1" date="1708516800">- OpenSIPS minor stable release: 3.4.4-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.5-1" date="1713441600">- OpenSIPS minor stable release: 3.4.5-1</changelog>
</package>
<package pkgid="d4a31386682630dcb82d9cfc5c548c630badcda35f2c205001223663e179c4ee" name="opensips" arch="src">
  <version epoch="0" ver="3.4.7" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-1" date="1689768000">- OpenSIPS 3.4.0 stable release: 3.4.0-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.4-1" date="1708516800">- OpenSIPS minor stable release: 3.4.4-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.5-1" date="1713441600">- OpenSIPS minor stable release: 3.4.5-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.6-1" date="1718798400">- OpenSIPS minor stable release: 3.4.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.7-1" date="1721822400">- OpenSIPS minor stable release: 3.4.7-1</changelog>
</package>
<package pkgid="c0e35c4b48776bfdc1b466df50bf556ac1becf9d03ae6c486b50f6e07be1646b" name="opensips" arch="src">
  <version epoch="0" ver="3.4.8" rel="1.el8"/>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.4-1" date="1708516800">- OpenSIPS minor stable release: 3.4.4-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.5-1" date="1713441600">- OpenSIPS minor stable release: 3.4.5-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.6-1" date="1718798400">- OpenSIPS minor stable release: 3.4.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.7-1" date="1721822400">- OpenSIPS minor stable release: 3.4.7-1</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.4.7-1" date="1724068800">- Replace deprecated dependency for radius modules</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.4.7-1" date="1724155200">- Add python-setuptools dependency</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.8-1" date="1724241600">- OpenSIPS minor stable release: 3.4.8-1</changelog>
</package>
<package pkgid="95f7d22ebc2623f0ceab6d9215f7d9cab7dddce44e9d7c45ec1dac5bc244f150" name="opensips" arch="src">
  <version epoch="0" ver="3.4.9" rel="1.el8"/>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.4-1" date="1708516800">- OpenSIPS minor stable release: 3.4.4-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.5-1" date="1713441600">- OpenSIPS minor stable release: 3.4.5-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.6-1" date="1718798400">- OpenSIPS minor stable release: 3.4.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.7-1" date="1721822400">- OpenSIPS minor stable release: 3.4.7-1</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.4.7-1" date="1724068800">- Replace deprecated dependency for radius modules</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.4.7-1" date="1724155200">- Add python-setuptools dependency</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.8-1" date="1724241600">- OpenSIPS minor stable release: 3.4.8-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.9-1" date="1729684800">- OpenSIPS minor stable release: 3.4.9-1</changelog>
</package>
<package pkgid="3d61f943d3713de34e0b0483b7f57c8ec9a66e060cfd3c7e03caf03d057333ee" name="opensips" arch="src">
  <version epoch="0" ver="3.5.0" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.6.0-1" date="1715256000">- Specification updated for opensips 3.6</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.5.0-1" date="1716033600">- Specification updated for opensips 3.5
- New modules: aka_av_diameter, auth_aka, mqueue, presence_reginfo, proto_ipsec, pua_reginfo</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.0-1" date="1721908800">- OpenSIPS minor stable release: 3.5.0-1</changelog>
</package>
<package pkgid="ba85883a3dbfd149bb918e27d3196919b561ff148d7479db524f5bf6844f4991" name="opensips" arch="src">
  <version epoch="0" ver="3.5.1" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.5.0-1" date="1716033600">- Specification updated for opensips 3.5
- New modules: aka_av_diameter, auth_aka, mqueue, presence_reginfo, proto_ipsec, pua_reginfo</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.0-1" date="1721908800">- OpenSIPS minor stable release: 3.5.0-1</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.5.0-1" date="1724068800">- Replace deprecated dependency for radius modules</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.1-1" date="1724241600">- OpenSIPS minor stable release: 3.5.1-1</changelog>
</package>
<package pkgid="09e3d54dd2741b59550f5ee6d0dbaf552eed96bce8129b2365b8b827b329ad33" name="opensips" arch="src">
  <version epoch="0" ver="3.5.2" rel="1.el8"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.5.0-1" date="1716033600">- Specification updated for opensips 3.5
- New modules: aka_av_diameter, auth_aka, mqueue, presence_reginfo, proto_ipsec, pua_reginfo</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.0-1" date="1721908800">- OpenSIPS minor stable release: 3.5.0-1</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.5.0-1" date="1724068800">- Replace deprecated dependency for radius modules</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.1-1" date="1724241600">- OpenSIPS minor stable release: 3.5.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.2-1" date="1729684800">- OpenSIPS minor stable release: 3.5.2-1</changelog>
</package>
<package pkgid="47c2c0cd346553553be06a49e7bb7e4e53c8a7b92a83bccc2424a026f3d96a8b" name="opensips" arch="src">
  <version epoch="0" ver="3.5.3" rel="1.el8"/>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.5.0-1" date="1716033600">- Specification updated for opensips 3.5
- New modules: aka_av_diameter, auth_aka, mqueue, presence_reginfo, proto_ipsec, pua_reginfo</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.0-1" date="1721908800">- OpenSIPS minor stable release: 3.5.0-1</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.5.0-1" date="1724068800">- Replace deprecated dependency for radius modules</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.1-1" date="1724241600">- OpenSIPS minor stable release: 3.5.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.2-1" date="1729684800">- OpenSIPS minor stable release: 3.5.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.3-1" date="1734523200">- OpenSIPS minor stable release: 3.5.3-1</changelog>
</package>
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  <version epoch="0" ver="0.11" rel="3.el8"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.8-3" date="1501156800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.8-4" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 0.9-1" date="1501761601">- Update to 0.9</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 0.10-1" date="1511438400">- Update to 0.10</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.10-2" date="1518091200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 0.10-3" date="1520596800">- Add gcc BR, spec cleanups</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.10-4" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 0.11" date="1537358400">- Update to 0.11</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.11-2" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.11-3" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
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  <version epoch="0" ver="0.12" rel="1.el8"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.8-4" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 0.9-1" date="1501761601">- Update to 0.9</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 0.10-1" date="1511438400">- Update to 0.10</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.10-2" date="1518091200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 0.10-3" date="1520596800">- Add gcc BR, spec cleanups</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.10-4" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 0.11" date="1537358400">- Update to 0.11</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.11-2" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.11-3" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.12-1" date="1593172800">- Update to release 0.12</changelog>
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  <version epoch="0" ver="0.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.2-1" date="1479988800">- First version of this spec file</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.2-1" date="1479988800">- First version of this spec file</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voidpomain.io&gt; - 0.2-2" date="1687953600">- Added patch with many improvements</changelog>
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  <version epoch="0" ver="2.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;ernaniaz@gmail.com&gt; - 2.1.0-1" date="1636027200">- First version for RHEL 8</changelog>
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  <changelog author="Ernani Azevedo &lt;ernaniaz@gmail.com&gt; - 3.0.0-1" date="1636027200">- First version for RHEL 8</changelog>
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  <changelog author="Ernani Azevedo &lt;ernaniaz@gmail.com&gt; - 0.1.2-1" date="1636027200">- First version for RHEL 8</changelog>
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  <changelog author="Kalev Lember &lt;kalevlember@gmail.com&gt; - 19-21" date="1430568000">- Rebuilt for GCC 5 C++11 ABI change</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 19-22" date="1434628800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-23" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-24" date="1486814400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-25" date="1501156800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-26" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-27" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-28" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-29" date="1549108800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 19-30" date="1564142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
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  <version epoch="0" ver="5.5.10" rel="4.el8"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 5.5.1-2" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Ha&#xEF;kel Gu&#xE9;mar &lt;hguemar@fedoraproject.org&gt; - 5.5.6-1" date="1540987200">- Upstream 5.5.6
- Fix issues with RHEL &gt; 7</changelog>
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  <changelog author="Bogdan Dobrelya &lt;bdobreli@redhat.com&gt; - 5.5.6-5" date="1550404800">- Revert use of systemd_ordering macro</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 5.5.6-6" date="1551960000">- Split off headless subpackage, based on idea from Bogdan Dobrelya</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 5.5.10-1" date="1551960001">- 5.5.10</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 5.5.10-2" date="1551960002">- Move reqs to headless</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 5.5.10-3" date="1551960003">- Move sysconfdirs to headless too</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 5.5.10-4" date="1552478400">- Minor clean up</changelog>
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  <version epoch="0" ver="0.1.3" rel="1.el8"/>
  <changelog author="Jiri Popelka &lt;jpopelka@redhat.com&gt; - 0.1-1" date="1405080000">- initial version</changelog>
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- Added gsl support</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.0-1" date="1450699200">- Updated to release 1.2.0</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.0-1" date="1477915200">- Updated to release 1.3.0</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1477915200">- Updated to release 1.3.1</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.0-1" date="1477915200">- Updated to release 1.4.0</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1482321600">- Updated to release 1.4.2</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1493208000">- Updated to release 1.4.3</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1637582400">- Updated to release 1.4.10</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545600">- Updated to release 1.4.5</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.0-1" date="1661947200">- Updated to release 1.6.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.7.0-1" date="1680264000">- Updated to release 1.7.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1703073600">- Updated to release 1.8.0</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1540987200">- Updated to release 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1605009600">- Updated to release 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1605009601">- Updated to release 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1621512000">- Updated to release 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1637582400">- Updated to release 1.4.10</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.7.0-1" date="1680264000">- Updated to release 1.7.0</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1605009600">- Updated to release 1.4.7</changelog>
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  <changelog author="Jiri Kucera &lt;jkucera@redhat.com&gt; - 14.4.2.0-20" date="1521633600">- added patch that fixes WAV to HCOM conversion abortion on 64 bit big endian
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- prevents division by zero in src/ao.c
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  + "invalid memory read in read_samples function in hcom.c" (CVE-2017-11358)
  + "divide by zero in wavwritehdr function in wav.c" (CVE-2017-11359)
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  <changelog author="Peter Oliver &lt;rpm@mavit.org.uk&gt; - 14.4.2.0-27" date="1577016000">- Package twolame-devel is absent from EPEL 8.</changelog>
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  <changelog author="Jiri Kucera &lt;jkucera@redhat.com&gt; - 14.4.2.0-29" date="1580299201">- backport fix of CVE-2017-18189 (rhbz#1545867)</changelog>
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  <changelog author="Igor Gnatenko &lt;i.gnatenko.brain@gmail.com&gt; - 0.0.6-1" date="1480334400">- Update to 0.0.6</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.0.6-5" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.0.6-6" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.0.6-7" date="1531569600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.0.6-8" date="1549195200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.0.6-9" date="1564142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
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  <changelog author="Prabin Kumar Datta &lt;prabindatta@fedoraproject.org&gt; - 0.1.0-1" date="1304510400">- Initial build</changelog>
  <changelog author="Prabin Kumar Datta &lt;prabindatta@fedoraproject.org&gt; - 0.1.1-1" date="1304510401">- upgraded to new version 0.1.1</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.1-2" date="1327492800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Hans de Goede &lt;j.w.r.degoede@gmail.com&gt; - 0.1.2-1" date="1347796800">- New upstream release 0.1.2
- Drop static lib
- Some spec-file cleanups</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.1.2-2" date="1409486400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Michael Kuhn &lt;suraia@ikkoku.de&gt; - 0.1.3-1" date="1444564800">- Update to 0.1.3.</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.1.rc1-1" date="1744459200">- Initial package file</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.1.rc1-1" date="1744459200">- Initial package file</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.1.rc2-1" date="1762430400">- Second release candidate</changelog>
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  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.3.0-4" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Richard Shaw &lt;hobbes1069@gmail.com&gt; - 0.5.1-1" date="1384430400">- Update to latest upstream release.</changelog>
  <changelog author="Petr Machata &lt;pmachata@redhat.com&gt; - 0.5.1-2" date="1400846400">- Rebuild for boost 1.55.0</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.5.1-3" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.5.1-4" date="1408363200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Petr Machata &lt;pmachata@redhat.com&gt; - 0.5.1-5" date="1422360000">- Rebuild for boost 1.57.0</changelog>
  <changelog author="Guido Grazioli &lt;guido.grazioli@gmail.com&gt; - 0.5.1-6" date="1424952000">- Rebuild for gcc switching default to -std=gnu11</changelog>
  <changelog author="Kalev Lember &lt;kalevlember@gmail.com&gt; - 0.5.1-7" date="1430568000">- Rebuilt for GCC 5 C++11 ABI change</changelog>
  <changelog author="Richard Shaw &lt;hobbes1069@gmail.com&gt; - 1:0.5.1-1" date="1517140800">- Revert epel7 branch back to 0.5.1.
- Add static library package.</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1:0.5.1-2" date="1563969600">- Make yaml-cpp compatible with boost 1.67+</changelog>
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